High-power floating point DSP technology ensures virtually unlimited internal dynamic range
Ultra high-resolution 24-bit AKM A/D and CRYSTAL D/A converters
12 ULN (Ultra Low-Noise) microphone inputs with analog inserts and switchable phantom power
4-band fully parametric equalizer, low-cut filter, gate, compressor and phase inverter on all 32 channels with additional delay function on the first 16 channels
Four built-in effects processors with first-class algorithms, such as reverb, chorus, flanger, phaser, delay, pitch shifter, tremolo, lo-fi, LFO filter, ring modulator and more
17 ultra-precise, low-noise 100-mm motorized ALPS faders
Freely configurable built-in level meters on all channels and channel controls with LED rings, which control any of nine selectable parameters per channel
Fully-featured static and dynamic automation functions
Four freely assignable analog outputs on balanced 1/4" TRS connectors
Internal input/output patchbay for easy routing of complex signal configurations
Six master controllers for comfortable push-and-turn functionality
Large, easy-to-read LCD display with adjustable contrast
Synchronization to SMPTE, MTC or internal clock
Dither, word length and noise shaping adjustable for digital main outputs
Two slots for installation of optionally available digital interfaces based on AES/EBU (8 I/O), ADAT (16 I/O) or TDIF formats (16 I/O)
MIDI and RS232 connectors allow communication with a PC or numerous other devices
Extensive MIDI implementation (MMC, program changes, control changes, MIDI sysex)
PC card slot for saving/loading various libraries and other settings
Free PC software for data transmission and management downloadable at www.ddx3216.com (serial cable included)
19" rack-mounting kit included
Manufactured under ISO9000 certified management system

BLOCK DIAGRAM

TABLE OF CONTENTS

1. INTRODUCTION... 10

1.1 Digital vs. analog..... 10 1.2 Concept..... 10 1.2.1 Architecture..... 10 1.2.2 Design concept and componentry.... 11 1.2.3 Open architecture.... 11 1.3 Before you begin..... 11 1.3.1 Shipment.... 11 1.3.2 Initial operation..... 11 1.3.3 Warranty..... 11
2. CONTROL ELEMENTS AND CONNECTORS... 12
2.1 Connection interfaces and control elements on the top side of the DDX3216... 12 2.1.1 Microphone and line inputs 1-12.... 12 2.1.2 Line inputs 13 to 16..... 12 2.1.3 Phantom power and 2-track inputs/outputs... 12 2.1.4 Control room and phones sections... 13 2.2 The rear panel of the DDX3216.... 13 2.2.1 Control room, multi and main outputs.... 13 2.2.2 Digital S/PDIF and wordclock inputs/outputs... 13 2.2.3 SMPTE and RS232 inputs.... 13 2.2.4 MIDI connections.... 14 2.2.5 Power supply and fuse..... 14 2.2.6 Option slots 1 and 2.... 14 2.3 PCMCIA card slot..... 14 2.4 Channels and main mix..... 14 2.4.1 Channel strips..... 14 2.4.2 Main Mix..... 15 2.5 Display..... 16 2.6 Snapshot automation: switches and displays... 16 2.7 Left switch block..... 16 2.7.1 Fader bank..... 16 2.7.2 Channel control bank.... 17 2.7.3 Proc(ess) bank..... 17 2.7.4 General bank..... 17 2.7.5 Auto(mation) bank..... 18

16. SPECIAL FUNCTIONS... 50
16.1 Updating the DDX3216 operating system.... 50 16.1.1 OS update with PC software.... 50 16.1.2 OS update with PC card.... 50 16.2 Recalling the factory presets and automatic fader calibration... 50

17. INSTALLATION... 51

17.1 Rack mounting..... 51 17.2 Audio connections..... 51 17.2.1 Analog connections.... 51 17.2.2 Digital connections (S/PDIF).... 51 17.3 MIDI..... 52

18. APPENDIX.... 53

18.1 MIDI Implementation..... 53 18.2 MIDI Control Changes..... 54
19. SPECIFICATIONS... 55 20. WARRANTY... 56

1. INTRODUCTION

Thank you very much for expressing your confidence in BEHRINGER products by purchasing the DDX3216. The BEHRINGER DDX3216 is an enormously powerful and fullfeatured digital mixing console based on 24-bit technology. Despite its compact size, the standard version of this console offers 16 complete inputs, four aux sends, four effect sends, 16 busses, four on-board effects processors and comprehensive routing options. Optionally available expansion modules (AES/EBU, ADAT and TDIF) allow you to upgrade your DDX3216 with 32 digital inputs and 32 digital outputs, so as to connect four digital 8-track recorders or 24-track hard-disk recording systems. The twelve ultra low-noise microphone preamplifiers featuring 24-bit CRYSTAL A/D converters can be enhanced by means of additional ADAT or TDIF-compliant 8-channel A/D converters to make sure that your DDX3216 provides enough connection options even for major-scale live applications. What is more, the DDX3216 features static and dynamic automation functions (snapshots) to record parameter changes. Your DDX3216 is equipped with a large number of ergonomically placed control elements, which you can use to operate a variety of functions in a very intuitive way. For example, the console has 17 faders controlling the levels of 32 input channels, 16 master busses, four aux sends and four internal effect sends as well as eight effect returns from the on-board effects processors. The knob, or channel controller on each channel is even more flexible, as it is always assigned to the same channel as the fader below it, but it can control any of nine different parameters on that channelchannel pan or the send level for one of the four aux or four FX sends. The display with adjustable contrast gives you a clear picture of various functions, such as EQ, dynamics, routing, delay, etc. The six master controllers below the display control the parameters as shown in the display. To give you maximum flexibility and enable you to work quickly and intuitively with the DDX3216, we designed the user interface so that it resembles that of an analog console. The faders made by ALPS are motorized, i. e. are automatically set to the correct position. The channel controllers have LEDs around them showing their current position. The switches above the faders are always assigned to the same channel as their faders, and give you direct access to the solo, mute and select functions. In addition, there is even a dedicated automation switch per channel. So, even when the display is used for other control functions, up to six parameters can be controlled simultaneously using the rotary controls or master controllers. You will find that the user interface of your DDX3216 can be operated even more quickly than that of a huge studio console with hundreds of controls! 2.

2.1 Connection interfaces and control elements on the top side of the DDX3216
The connectors and controls for the analog inputs are located in the upper section of the DDX3216. Ex factory, the analog inputs are assigned to channels 1-16.
2.1.1 Microphone and line inputs 1-12
The inputs 1-12 are mic/line inputs and have analog insert points (ISR = Insert Send Return).
Fig. 2.2: Connectors and controls for inputs 13-16 LINE IN The line inputs are on balanced 1/4" TRS connectors. SIG and CLIP LEDs These LEDs monitor the analog signal level after the insert point. When gain is in center position, the SIG LED lights at approx. -36 dBu (unity gain) indicating the presence of audio. The CLIP LED lights at approx. +10 dBu (unity gain) and warns you of signal distortion. GAIN Use the GAIN control to adjust the line input signal gain. The setting range is from -20 dB to +20 dB.
Fig. 2.1: Connectors and controls of analog mic/line inputs MIC The microphone inputs are on balanced XLR connectors and feature a switchable phantom power supply for condenser mics (cf. chapter 2.1.3 Phantom power and 2-track inputs/outputs). LINE IN The line inputs are on balanced 1/4" TRS connectors and function in parallel to the microphone inputs. INSERT The insert points are on 1/4" TRS connectors (tip = send, i. e. connection with input of external device; ring = return, i. e. connection with output of external device; shaft = ground, cf. chapter 17.2.1 Analog connections). In this way, you can insert analog signal processing devices before the channels A/D converters. Use commercially available insert cables (1/4" TRS on 2 x 1/4" phone connectors) for this connection. Insert points are very useful when you need to process channel signals with dynamic processors or equalizers. The insert points can also be used as tape sends to a multi-track recorder. PAD This attenuation switch (PAD) for line-level signals (or microphones with very high output levels) reduces the input gain by 20 dB. SIG and CLIP LEDs These LEDs monitor the analog signal level after the insert point. When gain ist closed, the SIG LED lights at approx. -46 dBu (mic)/ -23 dBu (line) indicating the presence of audio. The CLIP LED lights at approx. 0 dBu (mic)/+23 dBu (line) and warns you of signal distortion.

On the S/PDIF page in the I/O menu, you can assign the inputs of channels 13/14 to the digital S/PDIF input. When S/PDIF is selected as the signal source for channels 13/14, the normal channel input signals are replaced by the signals present at the digital S/PDIF input.
2.1.3 Phantom power and 2-track inputs/outputs
Fig. 2.3: Phantom power and 2-track inputs/outputs The +48 V phantom power required for condenser mics can be activated separately for channels 1-6 and 7-12. The associated switches light up when phantom power is on. CH. 1-6 This switch activates the phantom power supply for microphone channels 1-6. CH. 7-12 This switch activates the phantom power supply for microphone channels 7-12.
Make sure that the CLIP LED does not light up.
GAIN Use the GAIN control to adjust the mic/line input signal gain. The setting range is from +10 dB to +60 dB for the XLR input, and
Please mute your audio system before you activate the phantom power supply, so as to prevent switchon thumps from being passed on to your monitor speakers and/or headphones.
TAPE IN These RCA connectors (nominal level: -10 dBV) are used to return the signals from a stereo master recorder.
2.2.2 Digital S/PDIF and wordclock inputs/outputs
Press the 2 TK TO CTRL R switch to monitor the TAPE INPUTS via the control room and/or phones outputs. Fig. 2.6: Digital S/PDIF and wordclock inputs/outputs DIGITAL COAXIAL OUT The digital coaxial output (RCA) provides the MAIN MIX signal in a digital S/PDIF format. The parameters word length and dither for the digital output can be adjusted on the S/PDIF page in the I/O menu. DIGITAL COAXIAL IN This RCA connector allows you to feed in S/PDIF signals, with a sampling rate between 32 and 50 kHz. The input is fitted with a sample rate converter, so as to be able to feed in digital signals with a sample rate other than that used by the DDX3216. The S/PDIF input can be routed exclusively to channels 13/14, replacing the input signal connected to these inputs (see S/PDIF in I/O menu). If the DDX3216 is operated via its digital connectors, all digital devices connected to the console must be synchronized to a common wordclock rate. With an (optionally available) I/O module installed and devices such as digital multi-track recorders connected via digital leads, one of these devices must be defined as the wordclock master providing the clock rate for all other units. For this purpose, the DDX3216 generates internal clock rates of 44.1 or 48 kHz. In slave mode the console can be clocked via its wordclock input or from an device connected to an I/O module. The wordclock signal source is adjusted on the FS CLOCK page in the SETUP menu. Wordclock signals are usually distributed in a network configuration, i. e. using 75- coaxial cables, BNC-T adapters and terminating resistors. WORDCLOCK OUT The word clock output is on a BNC connector and provides a wordclock signal with the sample rate used by the console (TTL level square wave). WORDCLOCK IN The wordclock Input is on a BNC coaxial connector and accepts wordclock signals between 40 and 50 kHz.

3.5.2 COMP(RESSOR) menu page
The compressor works like a conventional analog compressor, providing gain reduction above the threshold, but offers the flexibility and performance of a digital dynamics processor. Press the COMP(RESSOR) switch in the CHANNEL PROCESSING section of the switch block to display the COMP(RESSOR) menu page for the selected channel. To scroll through the available menu pages, simply press the COMP(RESSOR) switch repeatedly.

3.5 Dynamics processing

Each of the 32 input channels is equipped with a full-featured digital dynamics processor, which also includes compressor/ limiter and gate functions. Similar to the EQ section, you can compare two settings (A and B) on the COMP(PRESSOR) and GATE pages, and use a dynamics library that includes a wide variety of pre-programmed settings with descriptive names. Of course, you can also save and load your own presets.
Fig. 3.7: COMP(RESSOR) menu page On the COMP(RESSOR) menu page, master controller 1 controls the key signal, which determines the amount of gain reduction applied. The key signal is used both for the compressor and the gate. The default is SELF/SUM, i. e. the input signal is also the signal used to determine the gain reduction. When channels are paired (stereo), the dynamics sections of both channels are linked automatically, and their mix signal serves as the key signal for both of them. This is to guarantee an identical gain reduction in both channels and thus a stable stereo image (see chapter 3.5.3 Compressor parameters). Master controller 2 adjusts the THRESHOLD and the ATTACK time. Turn this controller to edit the values of the selected controls (surrounded by a dotted line), or press it to switch to another parameter. Similarly, master controller 3 governs the parameters compression RATIO and RELEASE time. Master controller 4 determines the makeup gain (GAIN) as well as the compression characteristic curve at the threshold (KNEE). Finally, master controller 5 performs the A/B function (see above), and master controller 6 switches the compressor IN or OUT. The graphic on the right side of the display shows the curves and meters for the entire dynamics processor, including the gate. Displayed are the compressor characteristic curve based on the adjusted parameters threshold, ratio, knee and gain as well as the gate parameters threshold and range. In particular, the GAIN controller can be adjusted conveniently with the graphic display, and also the position of the IN/OUT switches is shown clearly: when the compressor and gate are off, the display shows a 45 straight line (indicating no processing). The far right part of the display shows two level meters for the input level (I = Input) and the gain reduction applied (GR = Gain Reduction). The GR value is the sum of the compressor and gate gain reduction.

seen on the COMP(RESSOR) menu page: with a hard-knee setting the graphic display of the compressor curve shows a sharp angle at the threshold, with one of the five softer settings the curve becomes gradually flat. The key signal determines the amount of gain reduction. Normally, the key signal is the same signal that is being processed (or the mix signal of both channels in stereo-link mode). In certain applications, however, it can be helpful to use another signal to control the compressor. For example, you can duck a music signal as soon as the signal delivered by an announcement microphone exceeds a certain level. Another common use of the key input is to use an equalized version of the signal to make the compressor respond only to certain frequency ranges, for example to eliminate esses (de-essing). On the DDX3216 you can set the key input to SELF/SUM (stereo linked when channels are paired) or assign it to any channel from the same fader bank. The compressor on the MAIN output always works in stereo-link mode.
3.5.4 DYNAMICS LIBRARY menu page
The DYNAMICS LIBRARY menu page offers a selection of pre-programmed dynamics processing presets with descriptive names. Of course, you can also name and save your own presets.
Fig. 3.8: DYNAMICS LIBRARY menu page To load a preset, turn one of the controllers below the preset list, until the preset of your choice is selected, then press RECALL. Loading a dynamics library preset will overwrite the A and B settings in the GATE and COMP displays (gate and compressor). The graphic display to the right of the preset list shows the compressor response curve of each preset to give you a quick overview of the parameter settings.

3.6 DELAY menu

Use the DELAY switch in the switch block to display a menu for the control of the channel delay and phase function. All of the 32 inputs feature a PHASE function, the first 16 inputs are equipped with an additional delay unit.
Fig. 3.9: DELAY menu Master controller 1 (PHASE) reverses the absolute polarity of the signal (display - = phase is reversed by 180). The DELAY section (inputs 1-16) provides a pure time delay for the alignment of signals in the time domain or to produce delay effects. Master controller 2 (FEEDBACK) routes the output signal from the delay unit back to the input, which allows you to create an echo effect. The higher the feedback value, the longer the echo duration. Positive settings produce in-phase feedback, and negative settings reversed-phase feedback. The DELAY controller determines the delay duration (0 to 276 ms), which is indicated in four different units of measurement (milliseconds (MS), distance in meters (M), samples (SMP) and beats per minute (BPM)).

Fig. 5.20: Parameters of Tremolo algorithm Wave (Sine/Tri/Squ) The Wave parameter defines the type of the modulating oscillation (sine, triangle, square). Speed (0.05 to 20 Hz) This parameter adjusts the frequency of the modulating oscillation and thus the speed of the amplitude modulation. Depth (0 to 100%) Depth governs the modulation depth, i. e. the amplitude of the modulating oscillation. High values produce a more pronounced effect.
Bass-Level (0 to 100%) The Bass-Level parameter defines the intensity of the harmonics generated for the low frequency range. NR-Response (20 to 400 ms) This parameter controls the response time of the gate. NR-Threshold (-90 to 0 dB) Use the NR-Threshold parameter to adjust the threshold below which the gate is activated.

5.2.24 Auto Filter

The Auto Filter algorithm is quite similar to the LFO Filter, but depends on the level of the signal to be processed. As this effect is actually an insert effect, we recommend that you route the effect post-fader, but disable the channels main routing. In this way, only the processed signal will be audible via the FX return path.

5.2.22 Graphic Equalizer

The Graphic Equalizer algorithm offers eight frequency bands, each with a maximum boost/cut of 15 dB. The following frequencies are available for processing: 50 Hz, 100 Hz, 250 Hz, 500 Hz, 1.5 kHz, 3.5 kHz, 7 kHz and 14 kHz. As this effect is actually an insert effect, we recommend that you route the effect post-fader, but disable the channels main routing. In this way, only the processed signal will be audible via the FX return path. Fig. 5.25: Parameters of Auto Filter algorithm Base Frequency (100 Hz to 10 kHz) This parameter defines the cutoff frequency of the filter. Filter-Mode (LP, HP or BP) Use this parameter to determine the type of the Auto Filter: LP (Low Pass), HP (High Pass) or BP (Band Pass). Sensitivity (0 to 100%) This parameter defines the filter sensitivity, i. e. the threshold above which the filter opens. Fig. 5.23: Parameters of Graphic Equalizer algorithm Filter-Q (1 to 20) This parameter controls the filter resonance and allows you to accentuate the frequencies close to the cutoff frequency. Attack (10 to 1,000 ms) Defines the time until the filter opens. Release (10 to 1,000 ms) Defines the time during which the filter closes.

Buzz-Gain (0 to 100%) The Buzz-Gain parameter determines the volume of the hum noise generated. Noise-LP (200 Hz to 20 kHz) This parameter adjusts the cutoff frequency of a low-pass filter, which is effective exclusively on the noise generated by the LoFi algorithm. Buzz-Freq (50/60 Hz) This parameter adjusts the frequency of the hum noise.
6. MONITOR SECTION AND LEVEL METERS
The DDX3216 gives you powerful monitoring features, such as a dedicated solo bus, which allows you to monitor the soloed PFL or AFL signal of the input channels or all aux and bus outputs. What is more, you can monitor almost any input or output via the control room and/or phones outputs. The control room output is usually connected to the monitoring system, and normally carries the stereo main mix or solo signals. Its outputs are on balanced 1/4" TRS connectors with a nominal level of +4 dBu, but can also be used with unbalanced connectors. The output level is controlled with the CONTROL ROOM LEVEL potentiometer. The MON -20 dB switch in the MAIN strip can be used to reduce the level at the control room or phones output by 20 dB.

5.2.26 Ring Modulator

This effect allows you to shape audio signals quite drastically. Similar to medium wave radio, the signal is multiplied by a carrier frequency. The resulting effect is ideal for robot voices, etc.
Fig. 5.27: Parameters of Ring Modulator algorithm Mod.-Mode (Tri, Sin, Squ or Env) This parameter controls the shape of the modulating waveform (triangle, sine, square or envelope). Mod.-Depth (0 to 100%) Mod.-Depth controls the modulation depth, i. e. the amplitude of the modulating oscillation. High values produce a more pronounced effect. LFO Speed (0.1 to 100 Hz) The LFO Speed parameter adjusts the speed (frequency) of the modulating signal. Env-Response (10 to 1,000 ms) Defines the envelope attack and release times. AM-Carrier-Freq (100 Hz to 10 kHz) Use this parameter to adjust the carrier frequency. AM-Depth (0 to 100%) The AM-Depth parameter determines the AM modulation depth. Bandlimit (100 Hz to 20 kHz) Use this parameter to adjust the cutoff frequency of a subsequent low-pass filter used to cut hard high-frequency content.

Please note that this switch is not effective on the 2-track signal.
The 2 TRACK TAPE IN connectors allow you to return the signals from a stereo recorder and play them back via the control room and/or phones outputs. They are on RCA connectors with a nominal level of -10 dBV. Enable the 2 TK TO CTRL R switch to monitor the 2 TRACK TAPE IN via the control room bus and phones outputs. The signal applied to the 2 TRACK IN can be routed to input channels 15/16 with the TO CH 15/16 switch. The PHONES output basically carries the same signal as the control room output. The headphones volume can be adjusted independently of the control room level with the associated LEVEL control.

6.1 MONITOR menu

Fig. 6.1: MONITOR menu Press the MONITOR switch in the left switch block to display the MONITOR menu, in which you can determine the signals to be sent to the control room and/or phones outputs. You can also adjust the solo level and set the solo function from PFL to AFL, and vice versa. PFL (pre-fader listen) is pre-fader and pan, AFL (after-fader listen) is after-fader and pan.
6.1.1 Mono/stereo switching
The left master controller in the MONITOR menu controls the mono/stereo switching. This function is particularly useful to position and align two stereo speakers (in MONO mode, the signal source must be located precisely between the two speakers), to isolate specific signals, or to check a mix for its mono compatibility. The following options are available: Stereo The selected signal source is monitored in stereo. Mono L The left channel of the selected signal source is sent at equal levels to both channels of the control room output. Mono R The right channel of the selected signal source is sent at equal levels to both channels of the control room output.
Mono L + R The left and right channels of the selected signal source are mixed together. The resulting signal is reduced in level by 3 dB to compensate for the higher loudness and avoid clipping. Subsequently, the signal is routed at equal levels to both channels of the control room output.
Solo for the aux, FX and bus outputs is always postfaderindependently of what has been set in the MONITOR menu.

6.1.2 Monitor switching

In addition to the stereo main mix or solo signals, any other inputs, aux, FX or bus outputs of the console can also be routed to the control room output. The signal to be monitored is selected in the center section of the MONITOR menu. Master controller 2 (MONITOR) switches the control room output between MAIN MIX and SELECT. Master controller 4 (SOURCE) determines the signal source to be monitored, when the SELECT position is selected. The inputs as well as the aux, FX and bus outputs are all displayed in groups of two. Use master controller 1 (MODE) in positions MONO L and MONO R to assign a single aux, FX or bus output to the control room output. In this way, you can monitor odd aux, FX or bus signals in position MONO L, and even aux, FX or bus signals in position MONO R.

9.2.5 Loading ALL files

Loading ALL files returns the console to almost the same state as when the file was saved, re-establishing all snapshots, libraries, user preferences, setup and dynamic automation settings. Any unused snapshot and library presets will be deleted.
9.2.1 Formatting a PC card
Before you can save data to a PC card, it must be formatted, which is done on the FORMAT page in the FILES menu.
Loading an ALL file overwrites the entire memory contents of your DDX3216, including all snapshots and libraries as well as all preference and dynamic automation settings!
9.2.6 Deleting snapshot or library presets
Fig. 9.4: FORMAT page in the FILES menu Master controller 6 formats the PC card inserted into the PC card slot. When CLEAR ALL is selected, all preset memories will be deleted (the on-screen switch changes to CLEAR). Pressing ENTER brings up a dialog, prompting you to confirm your deleteall command. Clearing the snapshot or library presets will irrevocably delete all of your USER presets!
Formatting will erase all data on the PC card.
9.2.2 Saving files to a PC card
After formatting, files can be stored on a PC card, using the PC CARD page in the FILES menu. Turn the master controller on the left (JOB), select SAVE and the file type of your choice (TYPE), and assign a name to the file. Press ENTER to save the file to the PC card. When you select the file type UPDATE!, the DDX3216 will store a copy of its current operating system on the PC card, which can then be used to update other DDX3216 consoles without a PC involved.
Fig. 9.5: PC CARD page in the FILES menu

10. SNAPSHOT AUTOMATION

The comprehensive snapshot automation of your DDX3216 stores almost all of the consoles settings in one of 128 preset memories. To save or load these presets, just a few operating steps are required, so that you can set up the entire console within a few seconds for a completely new mix or project! The serial port and the MIDI interface enable you to save your presets to a PC card or personal computer, for transfer to another DDX3216 or for backup and archiving. Snapshots can also be switched over by means of program change commands (see chapter 13.3 RX/TX page in the MIDI menu).
Fig. 10.1: SELECT page in the SNAPSHOT menu The FADE time can be adjusted with master controller 2. This parameter allows for a gradual crossfade between the current mixer settings and the stored snapshot settings, which includes all faders, pan controls as well as the aux and FX send levels. All other control elements, including the mute, channel routing and channel processing settings are loaded instantly. With the UNDO switch above master controller 6 you can undo a preset load command, and reset all control elements to their positions before loading a snapshot preset.

Fig. 14.1: INPUT page in the I/O menu The I/O menu controls the routing of the interface inputs and outputs. On the INPUT page (see chapter 8.2.1 Input routing), you can determine which interface input is assigned to which channel strip. This assignment is based on banks comprising 8 inputs each. For instance, you could assign inputs 1-8 of the interface to channel strips 1-8 on the DDX3216. Figure 14.1 shows a setup that has all inputs of the I/O modules assigned to channel strips 1-32. The MODULE page (see chapter 8.2.4 MODULE page in the I/O menu) assigns the 16 available outputs to the optional I/O modules.
Fig. 14.2: MODULE page in the I/O menu Each I/O modules is fitted with 8 or 16 outputs, and each group of 8 outputs can be assigned to the internal outputs 1-8, 9-16 or to the MULTI-MR outputs. Figure 14.2 shows one ADAT module installed in the first option slot, and one TDIF module in the second. I/O outputs 1-16 are specific outputs for the I/O modules, which can carry almost any mixer signal (OUTPUT page in the I/O menu). The default assignment is mix busses 1-16.

14.1 AES/EBU

The AES/EBU interface (Audio Engineering Society and European Broadcasting Union) is mainly used in professional studio environments and broadcasting applications for the transmission of digital signals over greater distances. The connections are made using balanced XLR cables with a specific cable resistance of 110 ohms. The cables may be 300-1500 feet long. With some minor adaptations even cable lengths of more than 0.5 miles are possible, which is not unusual in TV and broadcasting applications. The interface complies with the AES3 format, which allows for the two-channel transmission of signals with a resolution of up to 24 bits. The signals are auto-clock and auto-sync, which is important for the connection of multiple digital devices. So, there is no need for an additional wordclock connection between the DDX3216 and connected AES/EBU equipment. The sampling rate is not fixed, and can be selected freely. Typical sampling rates are 32 kHz, 44.1 kHz and 48 kHz. The AES/EBU interface is largely compatible with the S/PDIF interface widely used in semiprofessional applications. The connection can be made using a specific adapter. However, as there are some differences in terms of data structure and electrical specifications, we do not recommend the use of such a connection.
Type Connection Mode of operation Impedance Level Clock accuracy Jitter AES/EBU XLR balanced 110 ohms 0.2 V to 5 V pp not defined 20 ns IEC 958 (S/PDIF) RCA/optical unbalanced 75 ohms 0.2 V to 0.5 V pp I: 50 ppm II: 0.1 % III: variable pitch not defined
The DDX3216 can only transmit, but not receive word clock signals via its TDIF interface. For this reason, the DDX3216 must always be configured as the master. If you wish to define a connected TDIF device as master, you will need to establish an additional wordclock connection.

Please make sure that channels 1-16 have not MAIN activated in their routing, as this could lead to a superposition of signals, when combined with channels 17-32 carrying the tape return signals. When doing recordings, always monitor the tape return signal coming from the multi-track recorders!
Use the faders of channels 1-16 to adjust the levels of the digital recorder tracks, and the pre-fader aux sends of channels 17-32 to provide the monitor mix for the musicians. Connect your monitor system (e. g. BEHRINGER TRUTH B2031) to the control room outputs. The MAIN output provides the mix signal for the 2-track master recorder.
15.2 The DDX3216 in live applications
15.2.1 Live recording with the DDX3216
With its flexible routing options, the BEHRINGER DDX3216 is the ideal tool for live recordings, as you can create separate FOH and recording mixes with only one console! First, define the routing for channels 1-16 and 17-32. As the instruments connected to the analog inputs need to be available for recording as well, the analog inputs 1-16 are routed to channels 1-16 and 17-32 (INPUT page in I/O menu). This enables you to create an independent mix (including equalization and effects) with the faders of channels 17-32. In the ROUTING menu, channels 17-32 are sent to the 16 busses of the DDX3216, from where they are recorded on a multi-track recorder, using an optional digital interface. In this configuration, the level adjusted with the faders of channels 17-32 is independent of the level of channels 1-16. Such a routing configuration is also suitable for TV applications. For example, when doing a live TV show, the FOH and TV mixes are always separate, because the equalization of e. g. clip-on microphones must be adjusted differently for the P. A. and TV transmission systems, which is no problem when using the DDX3216, as this console can be used for both tasks.
As all console parameter can be frozen and recalled when required, you only need a few seconds to re-configure the entire equipment for a new band entering the stage. For example, in the run-up to the festival you could agree with each band on their specific setup, and can pre-program the console accordingly. Often, the EQ and effects library will be of valuable help to adjust some basic settings, which can be fine-tuned later on if need be.

16.1.2 OS update with PC card
Insert the PC card containing the new DDX3216 operating system into the PC card slot, then access the PC CARD page in the FILES menu.
Fig. 16.2: PC CARD page in the FILES menu

17. INSTALLATION

17.1 Rack mounting
The packaging of your DDX3216 contains two 19" rack mounts for installation on the side panels of the console. Before you can attach the rack mounts to the DDX3216, you need to remove the screws holding the left and right side panels. Then, use these screws to fasten the two rack mounts, each specifically to one side. With the rack mounts installed, you can mount the DDX3216 in a commercially available 19" rack. Be sure to allow for proper air flow around the unit, and do not place the DDX3216 close to radiators or power amps, so as to avoid overheating.
Fig. 17.3: 1/4" TRS connector
Only use the screws holding the DDX3216 side panels to fasten the 19" rack mounts.

17.2 Audio connections

17.2.1 Analog connections
You will need a large number of cables for the various connections of the console. The illustrations below show the wiring of these cables. Be sure to use only high-grade cables. Use commercial RCA cables to wire the 2-track inputs and outputs. You can also connect unbalanced devices to the balanced input/outputs. Use 1/4" phone connectors, or link the ring and shaft when using 1/4" TRS connectors (or pins 1&3 in the case of XLR connectors).
Fig. 17.4: Insert send return 1/4" TRS connector
Fig. 17.5: 1/4" TRS connector for headphones connection Fig. 17.1: XLR connections
17.2.2 Digital connections (S/PDIF)
Figure 17.6 shows you how to wire the unbalanced S/PDIF inputs and outputs by means of RCA connectors. Experience has shown that the type of cable used makes no difference. With lengths shorter than 30 feet, commercially available coaxial line cables are uncritical. For longer lengths or demanding applications, however, you should use cables having an appropriate cable resistance of 75 ohms.
Fig. 17.2: 1/4" phone connector
Fig. 17.6: Unbalanced connection (S/PDIF)

17.3 MIDI

The MIDI standard (Musical Instruments Digital Interface) was developed in the early 80s to enable electronic musical instruments of different brands to communicate with each other. Over the years, the range of MIDI applications has constantly been expanded, and today it is completely normal to network entire recording studios using the MIDI standard. At the heart of this network we find a computer loaded with a sequencing software that controls not only the keyboards, but also effects and other peripheral devices. Your DDX3216 can be integrated easily into such a studio environment. The MIDI connectors on the rear panel are internationally standardized 5-pin DIN jacks. To connect your DDX3216 to other MIDI equipment, you need dedicated MIDI cables, which are commercially available in various lengths. However, you can also use two-conductor shielded cables (e. g. microphone cables) and two rugged 180 DIN plugs to make your own MIDI cables: pin 2 (center) = shielding; pins 4&5 (right and left of pin 2) = internal conductor; pins 1&3 (the two outer pins) are not used. MIDI cables should not be longer than 50 feet.