SOME TERMS THAT WILL BE USED IN THIS MANUAL: Configuration: This refers to a particular group of settings, with the primary parameter being the crossover type. Program: A program is a configuration with either the factory default or a user-created name. Configuration and Program may be used interchangeably, though a configuration must be programmed for a program to exist. Parameter: A component of a setting such as the center frequency of one band of parametric EQ or the attack time of a limiter.
WARNING! To prevent possible speaker or
amplifier damage, always power-up peripheral devices first, wait 10 seconds, and then turn on the power amplifier. Turn off power amplifier first, then power-down peripheral devices.
WHAT IS A SPEAKER MANAGEMENT SYSTEM?
A Speaker Management System uses digital signal processing to accomplish multiple functions that affect the audio signal between the mixer and the power amplifiers. These functions include crossover, equalization, delay, limiting, high-pass and low-pass filtering as well as signal distribution. Programs may be created, selected and edited with the DS48's front panel controls. However, to create new programs, it is best to use the included free PC software and configure the programs using the graphical user interface (GUI) and a Windows PC.

2.1 Crossover:

This function uses filters to divide the audio spectrum into multiple bands, such as Highs, Mids, and Lows. When these separate signals are individually amplified and sent to a speaker matched for that range, the result is a much more efficient system with more punch and clarity as compared to system using a two or three-way speaker box with a passive cross-over. The common terms used for two or three-way systems are "bi-amp" and "tri-amp".

2.2 Equalization:

This function provides control of the tonal spectrum. It can be used to compensate for differing acoustic conditions, speaker cabinet responses or boosting/cutting certain frequency ranges to create a specific type of sound. In some rooms, "ringing" or feedback is present due to a number of conditions. Use the PEQ function to reduce the ringing frequencies.

2.3 Delay:

This function is used to "time-align" the different transducers in a speaker system. A properly aligned system will have better phase coherence resulting in more focused sound.
2.4 High Pass and Low Pass Filters:
These are used to reduce any signal outside of the desired bandwidth and are part of the equalization function. This frees the power amps and speakers from trying to reproduce signals that are not an intended part of the mix or simply beyond the reproduction range of the loudspeakers.

2.5 Limiting:

This function prevents a signal from exceeding a certain pre-determined level. It prevents input overload with its resultant distortion and also keeps amplifiers from clipping, which protects the speakers.

2.6 Signal Distribution:

This allows the signals from each of the four inputs to be processed and sent to the corresponding output. The crossover function is not active when in this mode. An LCRS (Left, Center, Right, SUB) mode is also available.

FRONT PANEL

1. LCD Display
The 2x20 backlit LCD displays the program and programming choices of the unit.
2. <BACK/NEXT> Buttons
These buttons allow navigation for selection of sub-menus and some parameter values.

3. Menu

This is one of two buttons that will access the front panel programming. This one is used primarily to select which program to call up, edit or save.

4. Gain

This is one of two buttons that will access the front panel programming. This, along with the <Back/Next> buttons accesses the various parameters of a program.

5. Enter

Hitting this button will write the current selection to memory.

6. Quit

This exits the programming menu and returns the display to showing the selected program.

7. FREQ Encoder Knob

This acts as the main value selector dial for most parameters other than those associated with EQ, Q and EQ Gain.

8. Q Encoder Knob

This adjusts the Q of a selected equalizer band.

9. GAIN Encoder Knob

This adjusts the Gain levels of Inputs, Outputs and any selected equalizer band.

10. Input Meters

These display the levels of the four inputs. The seven LED lamps per display are in 6 dB steps from 30dbV to 6dBV, then -3dBV, then Limit (0dBV), then Clip (+8 dBV)

11. Output Meters

These display the levels of the eight output channels. The seven LED lamps per display are in 6 dB steps from 30dbV to 6dBV, then -3dBV, then Limit (0dBV), then Clip (+8 dBV)

12. Output Mute Buttons

Pressing one of these buttons will mute the respective Output channel and the LED will illuminate. All outputs will mute automatically during certain program selection and editing processes, to protect the loudspeakers

REAR PANEL

1. Power Jack
This accepts a standard IEC type AC bayonet plug. Because the unit uses universal switching power supply, no voltage selection is required. The ON/OFF power switch is also located here.

2. RS232 Interface

Connect this to the serial port on a PC using the included cable or other standard DB-9 type cable. With the software installed, the PC can now monitor and program the DS48.

3. XLR Outputs 1-8

These are balanced (Pin 2 Hot) XLR outputs.

4. XLR Inputs A-D

These are balanced (Pin 2 Hot) XLR inputs that accept the signal from the outputs of the mixer.

QUICK START

NOTE: This section will describe navigation of the front panel controls as follows:
The button to press will be shown in ALL CAPITALS. For QUICK START, a number following in parentheses (#) will also be included. These refer to the same names and numbers as shown in Figure 1, Front Panel Controls and Indicators. The expected display text is shown in "quotation marks". If you press a button and the display does not change, it means that the button has no effect for that particular menu item. If you get to a certain point in the menu and want to go back, but cannot, press QUIT (6) and begin the steps again. This is normal for the unit.
1. Before hooking up the system, make sure that all items are OFF, especially the power amps. 2. Use high quality balanced XLR mic cables to make the connections. The feeds from the DS48 at "front of
house" can also be sent through the audio snake returns to the amp stacks if used in a live setting.
3. You should know the frequency response and limits of your loudspeakers 4. Connect the main output or outputs of your mixer to the inputs of the DS48. 5. Depending on the configuration you select, connect the appropriate number of outputs from the DS48 to your

power amp inputs.

6. Turn on the DS48. 7. After a brief start up, the display will show "2 x 2 WAY X-OVER" 8. Press MENU (3).

BACK button will not step you back, simply hit EXIT and start again. Where an additional "ENTER" press is required to write the selection to memory, the screen will prompt you with "[ENTER] to Confirm". The LED display also offers some navigation help. Left or right arrows on the top line indicate the BACK/NEXT buttons will step to other choices.
7.2 SUBMENUS 7.2.1.1 Crossover Submenu
This menu selects the type of crossover and is the primary submenu to create, recall, delete, and name a program configuration. Press MENU and ENTER to access this submenu. Then use the BACK, NEXT and ENTER buttons to adjust the Xover Submenu. Included submenus follow:

7.2.1.2 Load a Xover:

This will load a configuration that has previously been saved into the memory of the unit. Until you have saved other programs into the unit, nothing will be available to load. Only the previously selected program will display.

7.2.1.3 Design a Xover;

Use this mode to select from the ten different types of X-OVER and distribution configurations.

7.2.1.4 Store a Xover:

Store all the configuration settings of a X-OVER. The DS48 will store up to 30 program configurations in user memory locations. (An unlimited number can be stored on a PC or other PC accessible memory.) Each of the 30 programs can have a name containing up to 16 letters and numbers. If the program name is not edited, it is stored under the corresponding mode name, such "2 x 4 WAY XOVER".

7.2.1.5 Erase a Xover:

This will delete a user-created configuration.

7.3 Security Submenu

This menu allows you lockout various combinations of any or all front panel programming functions. A password with up to four-characters is created by the primary user and entered using the FREQ knob to select the character and the BACK/NEXT buttons to move the cursor to the four available spaces. To unlock the system, the password must be reentered.
Press MENU, NEXT and ENTER. Then use the NEXT/BACK and ENTER buttons to adjust the Security Submenu. The various locking modes are as follows: 1. LOCK: Changes only: Parameters can be viewed, but cannot be adjusted. MUTE buttons are valid. 2. LOCK: Changes + View: Parameters cannot be viewed or adjusted. MUTE buttons are valid. 3. LOCK: Changes + Mutes: Parameters can be viewed, but cannot be adjusted. MUTE buttons are not valid. 4. LOCK: Everything: Parameters cannot be viewed or adjusted. MUTE buttons are not valid. Once a security mode is selected, press ENTER. You will be prompted to enter a security code. This can be from one to four characters. Any of the characters available from the on-board selection choices may be used. A. Press the BACK/NEXT buttons to shift the cursor to one of the four positions. Then rotate the FREQ knob to select the desired character (letter, number or symbol). Repeat for the other three positions. B. Press ENTER, confirm the password and display as follows: C. The system is locked when the reentered password matches the original entry otherwise the locking operation is invalid. If invalid, the screen will show Bad Confirmation". If the codes match, the screen will show "Lock Success". Though up to four characters may be used, the lock out will work with just a single character. The initial and confirming entry must both be in the same position. NOTE: If nothing is entered (all blanks) the unit will still lock.

(The GAIN button will now select only the Output channels; the gain adjustment screens for the Inputs will not be displayed unless you use the BACK/NEXT buttons or exit the menu and start again.) Each Output will have both a number and name displayed. The names are taken from a default list with labels such as Low, Mid, Highs, Sub, etc. An alternate method is to program a single parameter for Output 1, such as DELAY, then press the GAIN button to go to each of the Output channels in turn and program the delay parameter for that channel. Now select the next parameter and program that for each of the outputs in sequence. In some cases this may make it easier to keep track of the comparative values between outputs. For simplification, we will follow though programming all the parameters for a single output. Set the Gain for Output 1. Range is 40 dB to + 6 dB

7.6.1.3 Polarity:

Set the polarity for Output 1. Switching this will change the phase of the signal 180 degrees. If Outputs are linked, reversing the polarity of one of the linked channels will also reverse the polarity of the other linked channel.

7.6.1.4 Delay:

Set the Delay time for Output 1. Delay is used to align drivers in a stack that may be in different vertical planes or multiple speaker in a stack. Delay range is 0ms to 7ms selectable in.1 ms increments. The delay is displayed in milliseconds, meters and feet. NOTE: Though there are some variables, generally speaking the speed of sound in 71 degree Fahrenheit dry air at sea level is ~773miles per hour. This is ~345 meters per second and about 1.13 feet per millisecond.5 milliseconds equates to about 6.78 inches.1 milliseconds equates to about 1.36. The DS48 displays 1ms = 343mm = 1.1ft.

7.6.1.5 HPF:

This is part of the crossover function. Sets the "knee" (corner frequency) and filter slope type for the HiPass Filter (HPF). Frequencies below the selected frequency will be filtered by the selected filter type and slope. Rotate the FREQ knob to select the frequency and the Q knob to select the filter type.

7.6.1.6 LPF:

Set the "knee" (corner frequency) and filter slope type for the LoPass Filter (LPF). Frequencies above the selected frequency will be filtered by the selected
filter type and slope. Rotate the FREQ knob to select the frequency and the Q knob to select the filter type. HiPass: frequency range 10Hz~16.0KHz LoPass: frequency range 35Hz~22.0KHz Use HPF and LPF together to define the frequency band for this channel of the crossover.

7.6.1.7 PEQ:

Set the parameters for the five available bands of parametric EQ. Use the FREQ, Q and GAIN knobs to adjust the three parameters in the display. These include Frequency, Q, and Gain (Cut or Boost). Pressing the ENTER button in this mode will select the symbol in the upper right corner. Depending on other settings, the choices will include some or all of the following: (PEQ), } (Lo shelf), { (Hi shelf), = (PEQ bypass) means PEQ means Lo shelf means Hi shelf means PEQ bypass Gain is +/- 12 dB

Input limiting Threshold

NOTE: when using Hi shelf or Lo shelf, set Gain=0.0dB, change Q-value to Hishelf/Loshelf.

} { =

The choices for the PEQ center frequencies are 360 ISO frequencies between 20 Hz and 20 kHz. Q is 0.5 to 128 and Hi Shelf or Lo Shelf.

7.6.1.8 Limiter:

The limiter is used to limit the signal going to the output stage and ultimately the amps, to a predetermined maximum level, with the intention of preventing clipping and its resultant distortion. There are four parameters available to set for the Limiter 1. Input Level sets the point at which the limiting circuitry will activate. This is sometimes referred to as Threshold. The range is -20dBV to +15dBV. A lower input level will activate the limiter sooner, affording more protection, but possibly creating more sonic artifacts if the limiter is activating and deactivating too frequently. Ideally, the limiter should be set so that it activates occasionally on peaks in the program. This control is also tied to the front panel level display meter for its selected output. As you lower the Limiter input level, the sensitivity of the meter will increase. This means a lower input level will result in the same signal showing more level on the Output channel display. Ideally, the output level should just approach and occasionally cross into limiting.

The signal BEFORE gain control and AFTER limiter processing
2. Attack. This is the amount of time that elapses before the limiter reduces the gain to the program level after the threshold point has been exceeded by the incoming signal. Generally, set shorter attack times on outputs that are feeding mids and highs and longer attack times for lows and subs. The time should be chosen carefully as the danger of dynamics distortion usually increases with shorter attack times. The attack time of the DS48 can be set within a range of 1 to 100 milliseconds. 3. Hold determines how long the limiter's gain reduction stays in place, even after the input signal has fallen below the Threshold. Use this to smooth out "choppiness", "pumping" or other artifacts that may occur under certain settings and program material. The hold time of the DS48 can be set within a range of 0.0 to 100 milliseconds in 1ms increments. 4. Decay: This control determines the how quickly the limiter removes the gain reduction and returns the signal to unity gain once the input signal has fallen below threshold. In instances where the limiter is constantly and repeatedly activated, a longer Decay time will prevent excessive retriggering and provide a more consistent signal. This control can be adjusted from 10ms to 1000ms in 10ms increments.

7.6.1.9 Input Signal:

In this menu, LCD displays the current input channel for the selected output. This parameter can be adjusted through all 4 input sources.
7.6.1.10 Name of channel:
This function allows you to choose a name or label for an output channel. There is a fixed list of 32 choices to choose from. Use the PARAM Rotate to change the channel name.
7.6.1.11 Copy Output Data
This allows you to copy the settings from one output channel to another, saving setup time. Press ENTER and select the source channel with either the FREQ knob or the BACK/NEXT buttons. Press ENTER again and select the target channel to copy to. Press ENTER and then ENTER again to confirm.

BLOCK DIAGRAMS

INPUT A
LOW OUTPUT 1 HIGH OUTPUT 1

INPUT B

LOW OUTPUT 2 HIGH OUTPUT 2

INPUT C

LOW OUTPUT 3 HIGH OUTPUT 3

INPUT D

LOW OUTPUT 4 HIGH OUTPUT 4

DS48 4X2 WAY

LOW OUTPUT 1 MID OUTPUT 1 HIGH OUTPUT 1 LOW OUTPUT 2 MID OUTPUT 2 HIGH OUTPUT 2
AUX OUTPUT 1 AUX OUTPUT 2

DS48 2X3 WAY + AUX

HPF SUM A+B
MONO SUB OUTPUT 1 MONO SUB OUTPUT 2

2X3 + MONO SUB

GAIN HPF LPF PEQS Delay GAIN MUTE LOW OUTPUT 1 LOW MID OUTPUT 1
HIGH MID OUTPUT 1 HIGH OUTPUT 1
LOW OUTPUT 2 LOW MID OUTPUT 2
HIGH MID OUTPUT 2 HIGH OUTPUT 2

2X4 WAY CROSSOVER

LOW OUTPUT 1
LOW MID OUTPUT 1 HIGH MID OUTPUT 1 HIGH OUTPUT 1

SUB OUTPUT 1

AUX OUTPUT 1
AUX OUTPUT 2 AUX OUTPUT 3

1X5 WAY AND 3 AUX

OUTPUT 1

OUTPUT 2

OUTPUT 3

HPF SUM A+B+C+D

OUTPUT 4

OUTPUT 5

OUTPUT 6

OUTPUT 7

MONO DISTRIBUTION

OUTPUT 8

LOW OUTPUT 1 LOW OUTPUT 2 LOW MID OUTPUT 1 LOW MID OUTPUT 2 HIGH MID OUTPUT 1 HIGH MID OUTPUT 2 HIGH OUTPUT 1 HIGH OUTPUT 2
HPF SUM A+B INPUT B GAIN HPF

HPF SUM C+D

STEREO DISTRIBUTION
HPF SUM A+B+C+D HPF INPUT C GAIN

LOW OUTPUT 5

LOW MID OUTPUT 6
HIGH MID OUTPUT 7 HIGH OUTPUT 8

LCR WITH MONO SUBS

MUTED OUTPUT 5

MUTED OUTPUT 6

MUTED OUTPUT 7 MUTED OUTPUT 8

4X4 PROCESSOR

MUTE OUTPUT 1 MUTE OUTPUT 2 MUTE OUTPUT 3 MUTE OUTPUT 4 MUTE OUTPUT 5 MUTE OUTPUT 6
MUTE OUTPUT 7 MUTE OUTPUT 8

MUTED, FLAT STARTUP

GRAPHICAL USER INTERFACE (GUI)
The BBE SYSOMAX software provides a graphical method of creating and editing programs on the DS48. It can also be used to make changes in real time, if adjustments are required during live operation. Full metering and control of all settings is available on the PC. Because it provides a more graphic and overall view, this method will allow you to program settings more efficiently than using the front panel navigation method on the DS48. Though the DS48 stores up to 30 programs, a virtually unlimited number can be written and stored on a computer with the software installed. Programs can be created whether or not the computer is connected to the DS48 processor. However, it must be connected to write or modify programs to the on-board memory of the DS48 itself. All the functions available onboard the unit are duplicated in the software.

8.1 MINIMUM SYSTEM REQUIREMENTS
Your SYSOMAX software has the following minimum system requirements for operation: Microsoft Windows 98, ME, NT4, 2000, or XP. Personal computer with Pentium 2 or higher processor. 20 megabytes (MB) of free hard disk space. 64 MB RAM. Computer RS-232 serial port for connecting the DB-9 type serial programming cable to the computer (if your computer only has USB ports, adapters are commonly available.) DB-9 type serial cable for connecting the device to the computer. This cable is included with the unit. Should it become lost or damaged, replacement cables are available at electronics and computer supply stores.
8.2 INSTALLING SYSOMAX ON YOUR COMPUTER
To install SYSOMAX from the CD-ROM: 1. Insert the CD-ROM into the computers CD-ROM drive. 2. Select My Computer from the Desktop or Start menu.
3. Click on BBE (D) (CD-ROM drive is usually drive D) 4. Double click on the file that then displays: DS48Setup_v13.exe 5. The Setup Wizard screen will appear. 6. Follow the prompts displayed on the screen to install the program. 7. When the installation is complete, eject the CD-ROM

8.3 STARTING UP SYSOMAX

To create a preset program on a DS48 unit , first connect it to the computer with the serial cable. If your computer only has USB ports, adapters are available from computer suppliers. NOTE: Alternatively, you can start SYSOMAX and create a program offline, without connecting to the unit. Always remember to save your programs. To upload the program to the unit, you must then connect the cable between the unit and the computer. After the connection is made you can start up SYSOMAX by using the Start menu on your Windows desktop, or by using Windows Explorer or My Computer.
8.3.1 From the Start menu:
1. On the Windows taskbar, click the Start button. 2. From the Start menu, select Programs, then DS48. Then click on the next DS48 icon that appears.
8.3.2 From Windows Explorer or My Computer:
Use Windows Explorer or My Computer to navigate to the folder in which SYSOMAX resides, then double-click the SYSOMAX.exe icon, or highlight it and press ENTER. Once you have started the program, you can choose to either operate the program online or offline to the unit. To exit the program, click the SYSOMAX Exit button in File menu of the title bar.

8.4 Navigating SYSOMAX

1. When the program starts, the main screen will be displayed. This is a flow chart of the various signal paths and processes. 2. You can select items from the drop down menus, the icons on the Toolbar or by clicking on one of the flow chart boxes.

8.4.1 Top Line Menu

This section provides an overview of the menu and screen choices. Those that require a more detailed description are covered in subsequent sections. 1. Exit lets you select EXIT to exit the program. 2. Config provides six choices: A. Online, which will open the RS-232 connection to the DS48 when its properly connected. B. Offline, which will close the connection to the unit. C. RS-232 lets you select the Com port for your particular computer. D. Config opens the Config window. This window lets you select one of the ten available modes. E. Data Copy lets you copy settings from one port to another, saving repetitive programming steps. F. Lock/Unlock opens the Lock Unit window. 3. Program offers two choices: Device Program Manager which opens the Memory Screen (see below) and Upload Current Program from Device, which will load the program currently running on the DS-48 to the computer for editing and/or saving. 4. View lets you select whether or not to display the Toolbar and Status Bar on your screen. The Toolbar is comprised of five Icons. 5. Help displays the version of the software. 6. Icon Bar offers five choices. From left to right they are : A. M opens the Memory management window. This is where you can name, save, erase, upload and download programs to.from the compute and the DS48. {See 8.4.2} B. RS232 Config C. Online/Offline D. Upload Current E. About
Device Program Manage/Memory
Clicking the Memory icon or selecting Device Program Manage form the Program Menu will open a dialog box with a table displaying 30 memory locations and spaces for names as well as 10 buttons. To Store a program as a memory location, click on the desired Prog line in the table for the desired save location. Then click either Store to Device or Store to PC. Write a new name in text box if desired and click OK. The dialog box will close and you will be returned to the main screen view. Refresh will upload the stored programs in memory from the device to the computer. To rename a program that has already been named, click on a program location in the table, then click Rename. Enter the desired name in the text box and click OK. The dialog box will close and you will be returned to the main screen view. Recall downloads the current version of the selected program from the computer to the device. Erase will immediately erase a selected program. Use caution as there is no confirm window before the command executes.
Erase All will erase all programs in the memory table and in the device. There is a confirm screen before this command executes. Exit will close the window.

Lock Unit

Found under config dropdown menu
Provides four options for security lockouts in the pull-down menu. Make your selection and enter your password in the text box. You will then need to confirm the password and OK the lockout. To unlock the unit, under the Config menu, click on Lock/ Unlock Unit. Enter the password in the text box and click OK. Then click on the Clear Password box. NOTE: If you forget the password, there is a code to unlock the unit so you can reset the password: COLO

9.6 Parametric Equalizers:
The next set of text boxes provides Frequency, Gain and Q setting choices for five bands of parametric equalization (PEQ) per output channel. There is also a check box for Flat, which will bypass the settings for that particular filter. In a similar manner to the Crossover filters above, clicking on any text box will open the PEQ filter settings window. Use the sliders to select values for Frequency Q and Gain (cut/boost). An additional check box for FLAT will bypass all value selections. At the bottom of the screen are forward and back arrows to step through the five PEQs. Gain must be set to 0 (zero) to select a different type of filter including the Hi or Low Shelf filter. Gain must remain at 0 for the shelving filters. Click the Set 0dB button to move the Gain slider to 0 dB or move the slider. If Low Shelf is selected and the Gain is moved to a setting other than 0, the shelving filter will be disabled and the screen functions will grey out. To adjust the gain for Input A, use your pointing device (mouse, touchpad, etc.) to position the cursor over the box. NOTE: Whatever program settings are current will be what is saved as the program, unless a different program is recalled.

SPECIFICATIONS

BALANCED INPUTS As standard, the DS48 comes with electronically servo-balanced inputs. This circuit design features automatic hum and noise reduction for balanced signals and thus allows for trouble-free operation even at high operating levels. Externally induced mains hum etc. will be effectively suppressed. The automatic servo-function recognizes the presence of unbalanced connectors and adjusts the nominal level internally to avoid level differences between the input and output signals (correction 6 dB).
Input Impedance 10KOhms CMRR 50dB (30Hz~20KHz) Output Impedance <50Ohms Maximum output electrical level Vpp=7.6V Frequency Resp 20Hz~20.0KHz Dynamic Range 110dB (A weighted) Distortion 0.01%(THD) A/D Processor AK5392; 24 bit data channel, processing Full 48-bit double-precision mode, 24bit/48khz sample rate AD latency 0.806MS; DA latency 0.513MS Maximum Delay 7ms (selectable in.5 ms increments) Output gain -40dB~+6dB (+/-0.5dB) Input gain -40dB~+6dB (+/-0.5dB) Parametric Equalizer Filter Gain +/-12dB in 1dB steps Center Freq 20Hz~20KHz 31 ISO Q value 0.5~10 Shelving Lo-shelf 20Hz~1kHz Hi-shelf 1kHz~20kHz Shelf gain +/-12dB in 1 dB steps HPF&LPF Filter Frequency (high pass) <10Hz~16.0kHz Frequency (low pass) 35Hz~22.0kHz Response curve Butterworth 6dB, 12dB, 18dB, 24dB, 48dB Bessel 12dB, 18dB, 24dB, 48dB Linkwitz-Riley 24dB, 48dB Limiter Level -40~+3dB Attack time 1~100ms Holding time 0~100ms Decay time 10-1000ms LCD 2X20 Backlit Input LED -30dB, -24dB, -12dB, -6dB, -3dB, Limit, Clip Output LED -30dB, -24dB, -12dB, -6dB, -3dB, Limit, Clip Connectors Input XLR-3F (Pin 2 Hot) Output XLR-3M (Pin 2 Hot) RS232 Power 110VAC 60Hz, 220VAC 50Hz, Weight 8 lbs (3.6 kg) Size 19 x 1.74 x 8.66 (80mmX44mmX220mm)

5381 Production Drive Huntington Beach, CA 92649 714-897-6766 FAX 714-896-0736 www.bbesound.com
BBE is the registered trademark of BBE Sound, Inc.

rev. 1 3/2006

P RO F ESS I ON A L
All the sound youve never heard

BBE Sonic Maximizer

What it is and what it does.
When we listen to live music, all of the highs and lows reach our ears in the same relationship to each other as when they were created by the instruments. If this same live music were to be recorded and played back through a loudspeaker system, the loudspeaker would introduce frequency-dependent phase shifting. The inductance of the speakers voice coil creates a stronger impedance as the signals frequency increases, resulting in a time delay. Consequently, frequency components with large negative phase shifts (high frequencies) arrive at the listeners ear later than signals undergoing small phase shifts (low frequencies). The resultant signal is distorted in the time domain to the listeners ear. Audio material containing sharp transients (e.g., percussive and plucked sounds such as drums, guitar, piano and harpsichord, etc.) suffers the most from this phenomenon, making it seem unfocused, or mushy. In order to address these problems inherent in basic loudspeaker design, BBE Sound, Inc. has developed a circuit that has two primary functions. The first adjusts the phase relationships
of the low, mid and high frequencies. Since a loudspeakers natural tendency is to add progressively longer delay times to higher frequencies, BBE sound technology adds progressively longer delay times to lower frequencies. This creates a kind of mirror curve to the time delay curve created by the speaker, neutralizing its phase distortion. The second major element in BBEs sound technology is the augmentation of the higher and lower frequencies. Loudspeakers tend to be less efficient in their extreme treble and bass ranges. Most sound-reproducing systems include a circuit for boosting high and low frequencies, showing an accepted awareness of the loudspeakers efficiency problem. The BBE system, however, provides a dynamic, program-driven augmentation which combines with the phase compensation feature to restore the brilliance and clarity of the original live sound. The result is, as one professional journal phrased it, The most hearable advance in audio technology since high fidelity itself!

Envelope Distortion

LIVE Original live sound before amplification. First part of transient shown.
BBE OFF Same sound as reproduced by an audio system. Note initial transient lost (envelope distortion).
BBE ON Same sound reproduced with BBE processing: corrects distortion; corrects phase delay.

Sonic Maximizers

The 882i is a dual-mono Sonic Maximizer that features our fourth generation processing engine and perfectly suited for any studio, PA system, guitar/bass rig or any application that requires balanced connectivity. Features include: A fullfunction Sonic Maximizer, +23dB of output headroom, balanced or unbalanced I/O, 115dB signal to noise ratio and Hard-wire bypass.
Frequency repsonse Process mode: program controlled Bypass mode: 10Hz to 50kHz, 0.5dBu, 0dBu input Signal to noise ratio 115dB Total harmonic distortion (THD) Process mode: less than 0.025% 20-20kHz Bypass mode: less than 0.002% 20-20kHz Maximum output +23dBu Input impedance 14.7K Ohms Output impedance 600 Ohms Sensitivity -45dBu for maximum process Maximum process +12dBu boost at 5kHz Lo Contour +12dBu adjustment at 50Hz, 0dBu input

The 482i is a dual-mono Sonic Maximizer that features our fourth generation processing engine and perfectly suited for any semi-pro studio, PA system guitar/bass rig or unbalanced/-10dB input headroom signal path. Features include: A fullfunction Sonic Maximizer, +18dB of output headroom, unbalanced "/RCA I/Os, 112dB signal to noise ratio and Hard-wire bypass.
Frequency repsonse Process mode: program controlled Bypass mode: 10Hz to 50kHz, 0.5dBu, 10dBu input Signal to noise ratio 112dB Total harmonic distortion (THD) Process mode: less than 0.025% 20-20kHz Bypass mode: less than 0.002% 20-20kHz Maximum output +21dBu Input impedance 47K Ohms Output impedance 1K Ohms Sensitivity -45dBu for maximum process Maximum process +12dBu boost at 5kHz Lo Contour +12dBu adjustment at 50Hz

382iSW

Now with 4th-generation processing. Features include: Sub Woofer Control with a variable low-pass filter (30-120Hz), full-function Sonic Maximizer, unbalanced "/RCA I/ Os, 108dB signal to noise ratio and Hard-wire bypass.
Frequency repsonse Process mode: program controlled Bypass mode: 5Hz to 30kHz Signal to noise ratio 108dBu process in 120dBu process out Total harmonic distortion (THD) Process mode: less than 0.1% at 1kHz Maximum output +16dBu Input impedance 47K Ohms Nominal input level -10dBu Output impedance 1K Ohms Nominal output level +16dBu Low-pass filter range 30-120Hz (variable)
The 382iSW combines a full-featured Sonic Maximizer engine with an independent variable subwoofer low-pass filter output. Applications for this versatile unit include: mobile DJ rigs, nightclubs/theaters, studios and PA systems.
Now with 4th-generation processing. Features include: A full-function Sonic Maximizer, +16dB of output headroom, unbalanced "/RCA I/ Os, 108dB signal to noise ratio and Hard-wire bypass.
Frequency repsonse Process mode: program controlled Bypass mode: 5Hz to 30kHz Signal to noise ratio 108dBu process in 120dBu process out Total harmonic distortion (THD) Process mode: less than 0.1% at 1kHz Maximum output +16dBu Input impedance 47K Ohms Nominal input level -10dBu Output impedance 1K Ohms Nominal output level +16dBu
The 382i series delivers all the sound improving benefits of a Sonic Maximizer with the convenience of ganged-stereo operation, perfectly suited for any unbalanced studio, PA system or guitar/bass rig application.

Compressors/Crossovers

MaxCom
The MaxCom is a professional and easy to use Dual channel Compressor/Limiter/ Gate and Sonic Maximizer that provides live or pre-recorded material with clean, distortion free compression. Whether the goal is to control vocals, a stereo mix or just dial in hard limiting to protect your speakers, the MaxCom will accomplish your needs with the reliability and sonic excellence that professionals count on. Features include: A full-function fourth-generation Sonic Maximizer, world-class THAT Corporation VCAs, balanced or unbalanced I/O, dual bar-graph meters for each channel to indicate Input Level and Gain Reduction and a Key in and out via " TRS.
Frequency repsonse 20 to 20,000Hz Signal to noise ratio More than 94dB Total harmonic distortion (THD) Less than 0.05% Threshold -40dBu to +20dBu Compression ratio 1:1 to 11:1 Attack time 0.1 to 200 msec Gate threshold Off to +10dBu

Max-X3

The Max-X3 is a professional 2-way stereo/3-way mono crossover and Sonic Maximizer, which offers a variety of options for any live sound application. The Max-X3 employs state-variable 4th-order Linkwitz-Riley filters to minimize phase inaccuracies in the crossover region, which is critical for the accurate summing of each driver in a speaker stack. Features include: 2-way stereo/3-way mono crossover, full-function fourth-generation Sonic Maximizer, Linkwitz-Riley filters (24dB per octave), Servo-balanced XLR inputs, " inputs/outputs, Low-cut subsonic filters for low frequency driver protection, phase inversion switching and over a 115 dB dynamic range.
Crossover type Stereo 2-way/mono 3-way Crossover frequency Low-mid range 1: 80Hz to 1kHz Mid-high range 10: 800Hz to 10kHz Filter type Linkwitz/Riley 24dB/octave Inputs Type: balanced XLR and unbalanced " Impedance: 100kOhms Outputs Type: balanced XLR and unbalanced " Impedance: 600 Ohms Hum and noise Low section (output @ 0dB): -106dBu High section (output @ 0dB): -97dBu Signal to noise ratio 118dB

Graphic Equalizers

EQA-231
The BBE EQA-231 is a professional dual 31band graphic equalizer. ideal for use in live sound rigs and installed setups such as studio, stage, DJ and sound contracting. The EQA-231 utilizes Constant-Q filters (constant bandwidth) this means that the bandwidth of each individual filter has a perfect balance between being narrow enough to prevent unwanted interaction between filters, yet wide enough to deliver exactly the type of correction curve for the ultimate in precision equalizing.

Constant-Q filters Variable low-cut filter from 10Hz to 250Hz (12dB/octave) Variable high-cut filter from 3kHz to 40Hz (12dB/octave) True hardwire bypass Balanced XLR or TRS input/output connections Clip indicator Line voltage switchable between 115 and 230 VAC operations

EQA-215

The BBE EQA-215 is a professional dual 15band graphic equalizer. ideal for use in live sound rigs and installed setups such as studio, stage, DJ and sound contracting. The EQA-215 utilizes Constant-Q filters (constant bandwidth) this means that the bandwidth of each individual filter has a perfect balance between being narrow enough to prevent unwanted interaction between filters, yet wide enough to deliver exactly the type of correction curve for the ultimate in precision equalizing.
Constant-Q filters True hardwire bypass Balanced XLR or TRS input/output connections Clip indicator Line voltage switchable between 115 and 230 VAC operations
Transients are enhanced, becoming more tactile, more present. Nuances in the high and low end are intensified; there is an added fullness and clarity. Stereo-program becomes more three dimensional, more spacious. Dense textures become more penetrable, with a more palpable space around each discrete instrument. This is powerful stuff. Maybe its voodoo? Home Studio Recording magazine
S p e a k e r M a n a geme n t
The DS48 loudspeaker management system features 4 inputs and 8 outputs with 10 configuration templates. Choice of operating and crossover modes includes: 4 x 2-way; 2 x 3-way + 2 aux; 2 x 4-way; 1 x 5-way + 3 aux, mono dist., stereo dist.; LCRS w/mono subs; 4 x 4 processor; and muted/flat startup. Some of BBEs DS48 features include high performance limiters on each output, providing complete control over attack, release and threshold parameters, variable high and low pass filters and your choice of 6dB, 12dB,18dB, 24dB, 48dB Butterworth, Linkwitz-Riley or Bessel frequency response curve. Independent control of each high and low pass filter allows asymmetric crossover bands to be created. The DS48 can also be controlled from a PC via the DS24s RS232 connection with the included software. Along with XLR input/output connectors, DS48 comes complete with selectable line voltages between 90 and 250 VAC operations.
24-bit AD/DA, 48k sampling rate Internal 32-bit DSP processing Parametric EQ provides -40dB to +6dB of gain with frequency, gain and bandwidth adjustment Full limiter for each output 7ms delay per output RS-232 computer control
The DS26 loudspeaker management system features 2 inputs and 6 outputs, and has a 5-band parametric equalizer for each output. Outputs all feature crossover filters, 5-band parametric equalizer, selectable high and low shelving filters, fully featured limiter and delay. Full metering is provided for inputs and outputs, with mute and access buttons allowing quick set up and gain adjustment. Other features included a choice of 12, 18 or 24dB/octave roll-off, and Butterworth, Bessel or Linkwitz-Riley responses. Independent control of each high and low pass filter allows asymmetric crossover bands to be created. The DS26 can also be controlled from a PC via the DS24s RS232 connection. Along with XLR input/output connectors, DS26 comes complete with selectable line voltages between 90 and 250 VAC operations.

As close as weve seen to a magic black box, the BBE Sonic Maximizer is probably the most cost-effective improvement you can add to your rig. Its a top contender for the Biggest Bang for the Buck award. Guitar Player

DI Solutions

DI-400
The DI-400 is a high-quality professional-grade fourchannel active DIbox for touring sound applications and recording studios. It has a major advantage over other high-end rack-mounted DIs on the market, with built-in Sonic Maximizer processor, on each channel, which employs the BBE 4th-generation processing engine just like the 482i and 882i Sonic Maximizer models. The DI-400 yields excellent results with active or passive bass guitars, keyboards, and both acoustic and electric guitars. The DI-400 is an excellent choice for transferring any musical instrument signal direct to a mixing console. An active direct box, its design is completely transformerless. Nothing to color your sound, its transparency is the DI-400s greatest attribute allowing the free expression of your instruments voice unfettered. The four-channel design, with rear inputs as well as front inputs, allows it to be permanently wired and rack-mounted in a touring rig or recording studio effects rack.
Frequency response @ line out Process mode: program controlled Bypass mode: 10Hz to 50kHz (1dB) Signal to noise ratio (per channel) BBE out: 118dB power supply (115dB 9v battery) BBE in (@10): 115dB power supply (112dB 9v battery) Total harmonic distortion (THD) BBE out: less than 0.005% at -10dBu input (20 to 20kHz) BBE in: 0.05% (20 to 20kHz) Noise BBE out: -105dBu BBE in (@0): -94dBu BBE in (@10): -85dBu Input impedance instrument: 1M Ohms line: 12k Ohms Output impedance xlr: 600 Ohms line: 1k Ohms (* 0dBu=0.775Vrms)

DI-100x

The DI-100x is a high-quality professional-grade active DI box designed primarily for recording applications, it was designed without compromise with paramount importance placed on maintaining the highest sonic integrity. Its major advantage over other high-end DI boxes on the market is its built-in Sonic Maximizer processor which employs the BBE 4th-generation processing enginejust like the 482i and 882i Sonic Maximizer models. Some of the DI-100xs features include: a full function BBE Sonic Maximizer; 15dB pad switch; ground lift switch; recessed switches to reduce the opportunities for damage; no-slip full bottom pad to stop slippage and prevent ground contamination through chassis to chassis contact; and a full 5-year warranty.
Frequency response @ line out Process mode: program controlled Bypass mode: 10Hz to 50kHz (1dB) Signal to noise ratio (per channel) BBE out: 118dB power supply (115dB 9v battery) BBE in (@10): 115dB power supply (112dB 9v battery) Total harmonic distortion (THD) BBE out: less than 0.005% at -10dBu input (20 to 20kHz) BBE in: 0.05% (20 to 20kHz) Noise BBE out: -105dBu BBE in (@0): -94dBu BBE in (@10): -85dBu Input impedance Instrument: 1M Ohms Line: 12k Ohms Output impedance XLR: 600 Ohms Line: 1k Ohms (* 0dBu=0.775Vrms) Power requirements DC inlet: +9 VDC min. 200ma Battery: 9 VDC 15 hour life

DI-1000

The DI-1000 is perfect for the recording studios and has no sonic equal in any other direct box currently available. The DI-1000 can operate as either a passive impedance matching device via the Jensen JT-DB-EP transformer, or as an active DI where the Sonic Maximizer circuit can be engaged with the signal still balanced at its output by the Jensen transformer. Some of the DI-1000s features include: a full function BBE Sonic Maximizer; Jensen JT-DB-EPC transformer; purely passive or completely active operation; switchable 0/15dB/30dB pad switch; ground lift switch; internal pure-silver wiring; recessed switches to reduce the opportunities for damage; no-slip full bottom pad to stop slippage and prevent ground contamination through chassis to chassis contact; and a full 5-year warranty.
Frequency response @ XLR balanced out Process mode: program controlled Bypass mode: 5Hz to 30kHz (1dB) Signal to noise ratio BBE out: 123dB power supply (118dB 9v battery) BBE in (@10): 118dB power supply (113dB 9v battery) Total harmonic distortion (THD) BBE out: less than 0.02% at -10dBu input (20 to 20kHz) BBE in: 0.1% at -10dBu input (20 to 20kHz) Noise Passive: -115dBu Active (BBE in @0): -110dBu Active (BBE in @10): -105dBu Input impedance Instrument: 1M Ohms Line: 12k Ohms Output impedance XLR: 600 Ohms Line: 1k Ohms Phase deviation Passive: less than 1 @ 20Hz, 0 @ 1kHz Active (BBE out): less than 11 @ 20Hz, 0 @ 1kHz (* 0dBu=0.775Vrms) Power requirements DC inlet: +9 VDC min. 200ma Battery: 9 VDC 15 hour life

Magnum D.I.

The Magnum D.I. is a professional active direct box housed in a rugged, ergonomic 14-gauge steel chassis, military-grade PCB, 2-way selectable pad (0dB/-20dB), ground lift, and can be powered by +48v Phantom power.
" input (unbalanced) thru (unbalanced) +48v Phantom power status LED 2-way input pad selector (0dB/-20dB) XLR output (balanced) Ground lift switch

Bench Press

Vintage Compressor The Bench Press is a vintage compressor that was patterned after the long discontinued Ross compressor. The heart of the Bench Press is fueled by a NOS CA3080 OTA (Operational Transconductance Amplifier) and 1% metal film resistors for better fidelity and features an Attack control to offer players more flexibility than the original.

Boosta Grande

Stomp Boxes
Boutique doesnt mean expensive.
Transparent clean boost The Boosta Grande is a transparent clean boost that is designed to produce up to 20dB of clean gain and benefit the player needing more volume or gain without changing their tone.
BBE stompboxes are not only 100% analog and designed in the USA, they are engineered and manufactured to specifically alleviate issues of reliability and inconsistency normally associated with vintage and homegrown manufactured effect pedals. Our pedals combine a boutique-style vibe with worldclass engineering, low tolerance/high quality components, true hardwire bypass switching and a five year warranty, all at a price that every musician can afford.

Crusher

High gain distortion The Crusher is a high gain distortion with a passive 3-band EQ (bass, mid and treble) that provides the player with a wide range of distortion effectsfrom a mild crunch to over-the-top, in your face distortion.
Features True Hard-wire bypass switching Road-worthy aluminum chassis High grade components Blue LED status indicator Easy access 9-volt battery compartment Non-slip rubber bottom Five year warranty

Free Fuzz

Seventies silicon transistor fuzz The Free Fuzz was patterned after a rare and sought-after silicon transistor fuzz pedal made famous by players like Jimi Hendrix in the late 60s. The Free Fuzz provides a wide range of creamy, dynamic fuzz effects with an almost endless amount of sustain for soloing and chunky rhythms.

Green Screamer

Vintage overdrive The Green Screamer is a vintage overdrive capable of producing a wide dynamic range of smooth and warm overdrive tones associated with vintage tube amps.

B B E S t o m p B o x es

Mind Bender
Dual-mode vibrato/chorus The Mind Bender is a dual-mode vibrato/ chorus that utilizes a BBD (Bucket Brigade Delay) circuit and produces a wide range of lush vibrato/chorus and rotating speaker effects.

Tremor

Dual-mode analog tremolo The Tremor is a dual-mode tremolo that utilizes an optical circuit to create a wide range of warm pulsating sounds normally associated with the vibrato effect of vintage tube amps. The Tremor offers the flexibility of two independent speed controls that are foot switchable to enable a player to select between a slow and a fast tremolo at the stomp of a switch.

Opto Stomp

Optical compressor The Opto Stomp is a transparent optical compressor that provides players with a wide range of soft-knee compression effects, reminiscent of the best vintage optical-element compressors made in the 60s. Perfect for use with bass and acoustic instruments.

Two Timer

Dual-mode analog delay The Two Timer was inspired by the long discontinued DM-2 Delay, which was known for its warm, haunting reverb and tape-like echoes. The Two Timer offers two independent delay times foot switchable via the time 1/time 2 mode switch. This makes it possible to have one delay set up for a short slapback rhythm sound while another is set to a longer delay for solos.

Sonic Stomp

Sonic Maximizer The Sonic Stomp is a stompbox version of our ever-popular Sonic Maximizer. The Sonic Stomp was carefully designed to deliver the same sonic improvement as our rack-mounted 482i Sonic Maximizer, adding clarity, definition and punch to any instrument.
Sixties wah-wah The Wah is based on a 1967 Class A circuit design that featured a unique Halo inductor. BBE has re-engineered and customtuned the Halo inductor and potentiometer to re-create the expressive tones made famous by players like Eric Clapton and Jimi Hendrix. In addition, weve added a harmony control to help fine-tune the Wahs Q/ peak response, and handwired all components are for the ultimate in vintage authenticity.

Soul Vibe

Sixties rotary speaker simulator The Soul Vibe is a vibe/rotary speaker simulator known for its chewy phase-like textures made popular in the late 60s and early 70s by such players as Jimi Hendrix, David Gilmour and Robin Trower.

Supa-Charger

Universal power supply The Supa-Charger is a high performance universal pedal power supply that features: a custom wound zero hum toroidal transformer; eight outputs that are isolated, regulated and filtered; and a selectable input mains switch to accommodate 110-220 voltages, enabling the Supa-Charger to be used worldwide.
I n s t r u me n t P r e a m p s

Acoustimax

BBE is at the forefront of audio technology which allows amplified music to sound more natural, and the latest generation of BBE High Definition Sound technology is a vital element to the new Acoustimax acoustic preamp. In the four years since BBE last offered its award-winning 386 acoustic preamp, BBEs design and engineering team studied the serious acoustic players needs for both recording and sound reinforcement. The result is Acoustimax, a foot-pedal-style (or stompbox-style) acoustic preamp which raises the bar for amplified acoustic instrument sound quality. Acoustimax is packed with features including three band, recording-console quality eq with a sweepable mid -range band, low frequency notch filter, phase reversal, ground lift, line level out, dedicated tuner out and TRS effects loop, even a pre/post switchable-balanced DI output with pad. Other features include hardwire bypass, non-slip rubber bottom and the included 12v power supply.

Frequency response 50Hz to 20kHz Treble 12dB @ 10kHz Mid 12dB @ 400kHz to 2.5kHz Bass 12dB @ 60kHz Notch 0 to -12dB @ 70Hz to 250kHz BBE Lo Contour: +15dB @ 50Hz Process: +15dB @ 10kHz Maximum output +10dBu @ instrument jack Noise Instrument jack: -98dBv (unity gain) Line out: -88dBv (+10dB gain) Total harmonic distortion (THD) 0.05% @ 1kHz Impedances Input: 2.2M Ohm Tuner out: 2k Ohm Line out: 1k Ohm Instrument out: 2k Ohm XLR out: 600 Ohm (* 0dBu=0.775Vrms) Power requirements DC inlet +12 VAC min. 200ma
The BMAX is a high quality bass preamp/ processor with applications both in the recording studio and on the road in demanding live applications. The BMAX has the warmth and dynamic range usually found in all-tube designs and delivers the punch and clarity of the best tube preamplifiers. At the heart of the BMAX is a class A/B front-end, the input of which emulates the loading effect that the input of classic-design tube preamp would present to the bass pickups, and one of the key aspects in the warmth and punch of tube preamps. The DI section is pure and clean with electronically- balanced output. 1% metal film resistors, military-spec PC board complete the no-compromise design of the BMAX.
Frequency response @ line out 15Hz to 30kHz 3dB Signal to noise ratio 100dB @ unity Total harmonic distortion (THD) Less than 0.02% Maximum output +22dBu Noise -77dBu @ unity gain, eq flat Input impedance Balanced XLR: 600 Ohms, pin #2 hot or + Unbalanced ": 1k Ohms Output impedance Balanced XLR: 600 Ohms, pin #2 hot or + Unbalanced ": 1k Ohms Interactive EQ Treble (boost only): 18dB @ 6kHz Mid (cut only): 12dB @ 300Hz Bass (boost only): 16dB @ 40Hz Bright switch +8dB @ 5kHz, eq flat Para-mids 12dB from 250Hz to 1kHz (* 0dBu=0.775mVrms)

BMAX-T

The BMAX-T is a high quality tube bass preamp/processor with applications both in the recording studio and on the road in demanding live applications. At the heart of the BMAX-T is a class A, DC regulated tube preamp section which follows in the tradition of the classic tube preamp designs of the 50s and 60s. The BMAX-T features a high voltage (300V) tube front end with hand-picked Groove Tubes 12AX7 tube, providing all of the pre-amp gain and passive interactive tone control. The high-end DI section is transformer-balanced with a genuine Jensen transformer for the ultimate in warmth and presence. 1% metal film resistors, military-spec PC board and custom-made toroidal power transformer complete the no-compromise design of the BMAX-T.
Frequency response @ line out 15Hz to 30kHz 3dB Signal to noise ratio 100dB @ unity Total harmonic distortion (THD) Tube section: 1% Post-tube section: less than 0.02% Maximum output +22dBu Noise -77dBu @ unity gain, eq flat Input impedance Passive: 1M Ohm Active: 100k Ohms Output impedance Balanced XLR: 600 Ohms, pin #2 hot or + Unbalanced : 1k Ohms Interactive EQ Treble (boost only): 15dB @ 6kHz Mid (cut only): 15dB @ 300Hz Bass (boost only): 15dB @ 40Hz Bright switch +10dB @ 10kHz, eq flat Para-mids 12dB from 250Hz to 1kHz (* 0dBu=0.775mVrms)

S o ft w a re

Sonic Sweet
BBE Sound, creators of the Sonic Maximizer and a leading manufacturer of professional and musical audio electronics, introduce the Sonic Sweet. The Sonic Sweet consists of three professional and easy to use Dynamics, Enhancement and Audio Enhancement plug-ins necessary for every facet of digital audio production. Whether youre tracking, editing, mixing or mastering, the BBE Sonic Sweet will give your projects a professional touch. Sonic Sweet supports sample rates up to 192kHz and is compatible with all AU, RTAS and VST host applications including ProTools, Logic, Cubase, Cakewalk and Digital Performer for Mac OS X, Windows 7, Vista and XP.
Operating system Windows XP Windows Vista 32/64-bit Windows 7 32/64-bit Mac OS X (10.4, 10.5 or later) Plug-in format Audio Unit RTAS/AS-Pro Tools 7 software or later VST 2.4 Sample rates Up to 192kHz sample rates supported

Stomp Ware

Stomp Ware is a suite of eight plugins modeled from our very own line of boutique analog stomp boxes. The suite consists of our Free Fuzz (70 fuzz), Green Screamer (vintage overdrive), Mind Bender (vibrato/chorus), Opto Stomp (optical compressor), Sonic Stomp (Sonic Maximizer), Soul Vibe (60 rotary speaker simulator), Tremor (vintage tremolo) and Two Timer (analog delay) to warm up every facet of digital audio production. Stomp Ware supports sample rates up to 192kHz and is compatible with all AU, RT AS and VST host applications including ProTools, Logic, Cubase, Cakewalk and Digital Performer for Mac OS X, Windows 7, Vista and XP.

The BBE Technology

Founded in 1985, BBE Sound, Inc. develops advanced audio enhancement technologies for not only the music and professional audio markets, but is also a global leader in high definition audio enhancement technologies for the Consumer Electronics marketplace. Manufacturers such as Panasonic, Clarion, Philips, Cowon, Sony, Hitachi, LG and many others proudly feature BBE technologies in products such as televisions, MP3 players, telephones, automobiles, and more. Look for the BBE Sound logo on all your Consumer Electronics products. It is the indicator of superior sound quality. I heard better dynamics and cleaner sound. restores a proper stereo imaging and separation. 15-20% increase in apparent openness and separation. the results were excellent. Stereophile

BBE Sound, Inc. 5381 Production Drive, Huntington Beach, CA 92649 Tel: 714 897-6766 Fax: 714 895-6728 www.bbesound.com
Recycled with 25% post-cosumer waste.