Behringer are known for making cost-effective products, but their new X32 live mixer takes that to a whole new level...
Formed in 1989, Behringer are one of the industry's best known 'pro audio' equipment manufacturers — almost everyone has used a Behringer product at some point! At the start of 2010, the Music Group, Behringer's parent company, acquired two closely linked British companies. Midas are very well known for class-leading live-sound mixing consoles, while Klark Teknik make ancillary equipment like graphic equalisers, system controllers, compressors and the classic DN780 reverb.
The three companies' joint technical expertise has now produced its first fruit in the X32, a product that looks set to completely overturn the market for budget digital mixing consoles. For anyone who hasn't already been caught up in the marketing excitement, the X32 is a fully featured, 32-channel, 16-bus digital mixer with a generous supply of built-in effects and an impressively comprehensive feature set. It is geared primarily towards live sound, but is sufficiently versatile to make studio applications perfectly viable too. Even more excitingly, it is set to sell at a street price of under £2000 in the UK. This dramatically undercuts the obvious competition, such as Yamaha's LS9 and Presonus's Studio Live, while competing on features with still more expensive desks such as the Soundcraft Vi series. The X32 has obvious appeal to the budget and semi-pro live-sound market, as well as small-scale pro users, theatres and houses of worship.
As with most digital desks, the X32's control surface belies its true power and potential. Behringer's design follows a similar paradigm to most digital consoles, where the physical inputs and outputs are connected, via a 168 x 168 digital routing matrix, to the internal digital signal processing core. The DSP offers a total of 40 input processing channels (32 mic channels plus six aux inputs and a stereo USB input), with 25 mix buses and output channels (including 16 internal buses, six matrix outputs and three main output buses).
The analogue inputs comprise 32 preamps (with individually switched phantom power) that present a relatively high 12kΩ input impedance and will happily accept line levels, plus six balanced auxiliary line inputs, two of which have alternative unbalanced RCA/phono connections. The preamp circuitry is similar to that employed in the Midas Pro 2, and significantly better than any previous Behringer offering.
Thirty-two channels of audio can be input from a computer via the XUF USB/Firewire interface card slot, and there are also two sets of 48 digital inputs on the desk itself. These use the AES50 'SuperMAC' standard — an ultra-low latency (typically 0.25ms) audio network hooked up with standard Cat 5 cabling — and are primarily intended for connecting Behringer's new S16 digital stage boxes. With up to three units per AES50 interface, a total of 96 sources and 48 returns is accessible from the console. The AES50 ports can also be used to share buses between X32 consoles.
An obvious omission is any form of conventional digital input, such as S/PDIF, which would enable direct connection of a CD player, for example. However, unbalanced analogue inputs are provided on auxes 5/6, and a USB memory stick port appears on aux inputs 7/8. Amusingly, the USB control window graphic mimics an old-fashioned cassette player, and the spools even rotate when playing! However, the USB replay facility doesn't recognise connected iPods or iPhones, or MP3 files at all. It will only play back linear WAV files, either directly from the root directory, or stored in folders, with options to play continuously through all files in a folder, or a single file at a time. Making it impossible to play low-rate MP3s over a big PA is probably a good idea, but it is bound to frustrate some! Also, the desk has to be set to the same sample rate as the files. This might cause frustration on occasion, but I'm told that there is a possibility of including sample-rate conversion in a v2.0 firmware.
On the analogue output side, the X32 features 16 freely assignable outputs on XLRs, plus a further six aux sends on TRS sockets (two with unbalanced RCA/phono sockets). There are also separate dedicated control-room/monitor outputs (paralleled on both XLR and TRS sockets), plus two headphone sockets buried in the hand-holds built into the X32's sides. A flightcase for the mixer would need to have edges low enough to not restrict access to these headphone sockets; it would also need to avoid impeding the (very quiet) slow-speed cooling fan on the bottom of the desk, which requires an unrestricted airflow.
Digital outputs include a solitary AES3 output, while aux outputs 7/8 are dedicated to the top-panel USB port to serve as a stereo record feed, creating an uncompressed WAV file automatically named with the starting date and time. A USB stick can also store console scene memories, presets and libraries. The XUF interface card provides 32 input and output channels to a computer via USB or Firewire, while the desk's two AES50 interfaces provide a duplicated set of 48 outputs. There is also a 'P16 Ultranet' interface for connection to Behringer's Powerplay P16M 16-channel personal monitoring system. The rear panel sports dual MIDI connectors, plus Ethernet and USB sockets to enable remote control using the planned XRemote software and iPad app (the latter requiring a Wi-Fi router to be connected).
The X32 ships with the XUF interface card installed as standard, with selectable USB2 (on the square type-B socket) and Firewire 400 connectivity — though only one can be active at a time. The interfaces are compatible with Core Audio for Mac OS 10.5 and higher, or with Windows via downloadable ASIO drivers. Separate drivers are provided for USB, Firewire, and MIDI (for controlling a DAW using the Mackie Control/HUI protocols). Using Firewire on a MacBook I was able to happily record and play back 32 channels with a latency below 10ms.
A Behringer representative told me that the XUF interface card was deliberately restricted to FW/USB connectivity with no frills, purely for the sake of development speed. However, more option cards will follow, and they are "currently evaluating the next steps to take depending on the popularity of other interface formats like MADI, Dante, Thunderbolt, and so on.” Given the console's strong live-sound emphasis, it would make sense for future interfaces to support common live-sound audio networking formats such as Cobranet, Ethersound, Dante, Ravenna and the IEEE AVB protocol. For the project studio market, multi-channel ADAT and AES3 interfaces will be a must — hopefully with word-clock I/O too. It would also be a welcome feature if the console could record its own mix buses or channel direct signals to an external drive, and I'm told there are plans for a card to enable this via USB.
Signal routing between the physical I/O and the DSP inputs and outputs is performed through a dedicated Routing menu page on the main display, where input sources are pre-allocated in blocks of eight. Each channel's specific input source can then be selected from the preamp configuration display page, with a wider range of sources that includes all the internal buses. Each of the physical outputs is assigned from a sub-page of the same menu, with options to select the type of source (main LCR output, mix bus, matrix, direct out or monitor), the specific output within that subset, and the 'tap' point (pre/post EQ or pre/post fader). There is also a variable delay feature for time-alignment applications (up to 500ms — the equivalent of 170 metres).
The DSP signal path for the 32 main input channels is very well featured, with polarity inversion, gain/attenuation (-12 to +60dB), delay (up to 500ms again), a variable high-pass filter, a gate/ducker, and an insert point, which can be assigned pre or post the EQ and compressor block. The inserts can be routed via the physical I/O, or to the internal effects engines. Next comes a versatile four-band equaliser and a compressor/expander, and their order can be reversed in the signal path. Both the gate and the compressor have filters on their key inputs. These can be set to high- or low-pass modes with 6 or 12 dB/octave slopes, or to band-pass response with adjustable bandwidth. The key input sources can be selected from any channel, aux input, effects return or mix bus.
The channel path continues with a channel mute, fader and panner. This is configurable depending on how you use the left, right and centre outputs, either for left/right stereo plus a separate mono send, or integrated left/centre/right panning. There's also comprehensive routing to the 16 mix buses, with the ability to take signals pre- or post-EQ and pre/post-fader. These 16 buses are used to create all the aux sends, audio subgroups, and any other mixed feeds required. The channel solo system is also configurable, with the monitoring signal being derived from any of the same points as the bus routing.
The aux line inputs have a simpler signal path, comprising just a ±12dB level trim, four-band EQ, mute, fader and pan, with similarly reduced mix-bus and solo-source options. The eight internal stereo effects return channels are even more basic, with just mute, fader, pan and bus routing.
There is a useful line-up oscillator which can output a sine wave or pink/white noise at adjustable level. A talkback section can use either the built-in mic or one plugged into a permanently phantom-powered top-panel XLR socket. There's a preset talkback compressor to maximise intelligibility; its output level is adjustable and the two talkback buttons are configurable for latch/momentary action. The oscillator and the dual talkback outputs can be routed independently to any of the mix buses and main outputs.
The DSP signal path for the three main output channels (left, right and centre) comprises an insert point, six-band EQ and compressor/expander. The order of the processing stages can be reversed, and the key input has its own filter. After this there's an On switch (instead of a mute) and a fader, before the signal reaches the physical outputs via the console's routing matrix. The post-fader signal can also be routed to any of the six matrix buses, or the solo bus.
Sensibly, the 16 submix buses enjoy exactly the same signal-path facilities, with the addition of selectable LR or LCR panning, while the six matrix mix buses also have the same signal-path structure but no panning facilities. Their individual outputs are available in the routing matrix. The monitor source can be selected from the LR mix buses, the stereo output pre/post the main fader, or directly from auxes 5-6 or 7-8 (USB). The selected source is overridden if a channel solo button is pressed. Dim and mono controls are provided, as well as a delay facility, and separate volume controls for the monitor and headphone outputs.
The eight assignable stereo/dual-channel effects engines offer a wide range of effects, including delay, chorus, phasing, flanging and pitch-shifting, as well as less common ones such as a transient enhancer, limiter, enhancer, guitar amp and tube preamp modeller, and rotary-speaker emulator. There is also a range of high-quality reverbs, and each engine can be switched to serve as a dual 31-band graphic EQ. The only obvious omission is a vocal de-esser, but Behringer promise new effects in future firmware updates (which will be free).
The first four effects engines can be configured either as conventional auxiliary effects for use via aux sends, or as inserts, while the last four can only be used as inserts. A typical configuration might involve four stereo multi-effects, fed from auxes with their returns feeding the mix buses, along with eight mono 31-band graphic EQs inserted into the monitor, matrix and FOH speaker feeds, for example.
Interestingly, the digital effects are claimed to be simulations of well-known devices such as the Lexicon 480L and PCM70, the EMT250, and Quantec QRS — although the Klark Teknik DN780 isn't listed! All sound very good indeed and are extremely usable. The graphical interfaces used for editing hint strongly at the inspirations, with Roland, Yamaha, EMT, Lexicon and other styles easily recognisable.
The X32 is a moderately sized desk, measuring 900 x 528 x 200mm, but weighs only 21kg, and the hand-holds in the side cheeks make it quite manageable by one person. The console consumes 120W and the integral power supply accommodates mains voltages from 100V to 240V AC. A four-pin XLR on the top panel provides 12V for a gooseneck lamp, if required.
The control surface employs the familiar approach of having an assignable 'channel strip' section across the top on the left-hand side, with a seven-inch, high-contrast 800 x 480-pixel colour TFT display and bargraph metering slightly offset from the centre. Monitoring and talkback facilities sit to the right of the display, with six 'soft' encoder/switches running across the bottom, while eight push-buttons along the right-hand side access various display and configuration screens. The top of every screen shows the currently selected channel, current and next scene number and name, the selected USB playback file name (with elapsed and remaining time) and the USB record status. Also displayed are the AES50, card slot and clock synchronisation status and sample rate, as well as the current time.
The lower half of the console boasts 16 input-channel faders, eight group faders, a main stereo output fader (all with 10dB of gain above the unity mark) and a section with freely assignable encoders and buttons, as well as six mute-group buttons, and scene recall facilities. There is also a handy rubber mat to house a smartphone or SPL meter.
The eight group faders are normally configured as assignable DCAs (digitally controlled amplifiers) or 'virtual groups'. Each DCA fader can control any number of other channel or group faders, without combining their audio signals first. Allocating channels to DCAs is achieved by holding down the DCA fader's Select button, then pressing the Select button from the relevant channel or submix. A similar assignment process is used for the mute group allocations, although a Mute Grp button must be selected first, to avoid accidentally muting channels when trying to add new ones to an existing group!
Every fader strip is equipped with an illuminated green Select button at the top, a simple bar-graph meter with an LED to indicate when the dynamics processors are functioning, a yellow Solo button, a 128 x 64-pixel colour LCD, and a red Mute button, with a motorised 100mm fader at the bottom. Four bank buttons switch these faders to control channels 1-16, channels 17-32, the eight aux line inputs or first four internal effects returns (with separate faders for the left and right channels), or the 16 submix buses. The eight group faders are similarly assignable to serve as eight DCAs, or to control the submix buses (in two banks of eight), or the six matrix output buses plus the centre/mono bus. The final fader always controls the main stereo outputs.
A Sends On Faders button between the two fader sections provides two ways of adjusting or checking each channel's sends to the submix buses. If a channel is selected on the left-hand fader set, the send level to each submix bus can be adjusted using the group faders (paged to access the two sets of eight). Alternatively, if a submix bus is selected on the right-hand fader set, the contribution from each channel to that bus can be adjusted with the channel faders.
Additionally, the eight group faders can be used in 'remote mode' to control a DAW via the HUI or Mackie protocols, with channel banking in blocks of eight. The 16 channel faders don't generate MIDI data for DAW control purposes, though. Although it appears that there are no dedicated DAW transport controls, when the Remote Layer function is enabled and selected, the LCDs above the group faders display all the usual functions, which are then activated with the mute buttons. The assignable section controls can also be told to send the appropriate MIDI messages, if required.
The colour LCD panel above each fader, where the user can select the background colour as well as a graphical symbol and an identifying name, is a very nice feature. Names can be selected from a fairly comprehensive preset list that spans most instruments, or a name can be entered manually. These settings can be saved as channel presets, and the last-used settings are automatically reloaded when the console is rebooted — something that takes a sprightly 15 seconds, by the way. The flexible colour-coding makes it very easy to identify related input channels, or to associate submix and DCA faders with specific groups of input channels.
To keep things simple and intuitive for a live-sound environment, only the most essential physical controls are provided, with additional but lesser-used facilities and displays accessible via the main screen and its 'soft' encoders. This means that the 'channel strip' panel (see photo elsewhere) is laid out quite spaciously, with 13 rotary controls (with LED collars) and 17 backlit buttons arranged in five sections.
The preamp configuration section comprises a gain control and a high-pass filter frequency knob; these can also control the remote preamps in the S16 stage box via the AES50 interface, if connected. Buttons select polarity reverse, phantom power and HPF on/off, with a red LED illuminating next to the XLR of any phantom-powered inputs. A button labelled View forces the main display screen to show these preamp settings in more detail (with numerical values for gain and HPF frequency), as well as providing an input-level meter, a facility to link adjacent odd/even channels, select the channel's input source, move the insert point pre- or post- the EQ/compressor block, and configure the insert signal routing. LCD scribble strip configuration is a hidden function, accessed with the Utility button alongside the main display.
This View button paradigm is used throughout to provide expanded facilities on the main display, so the channel strip's gate section has only a threshold control and an on/off button, but its associated display screen provides a 'transfer curve' graph and a gain-reduction meter, along with gate/duck mode selection, range, attack, hold and release controls in the first page, and key-input source, filter frequency and filter mode/Q controls on the second page. The Dynamics section is similar, but with ratio and gain controls instead of mode and range. (Unusually, there is no auto-release setting.) Where rapid and continuous access to functions is required, they can be allocated to the assignable controls (see below). Settings can also be saved and recalled to and from a library.
Comprehensive EQ controls are provided, with gain, frequency and Q controls alongside buttons to select the band, shelf, parametric or filter modes. The associated display screen shows a frequency response graph and numerical values for each parameter across four sub-pages, as well as a switch to reset the EQ flat. It would be nice if this function was available on the control surface — another firmware update, maybe? Again, pressing the Utility button accesses another page, with facilities to copy and paste EQ parameters between channels, and to save or recall settings from a library.
Finally, the bus routing section has controls to adjust the levels to four buses, with buttons to access the rest in groups of four, while the associated View screen displays numerical values and includes options to switch adjacent send pairs pre- or post-fader. Similarly, the main Output section has controls to adjust the level of the mono bus (and to turn it on or off), as well as the stereo bus pan and on/off status.
Scene navigation controls comprise Previous, Next, Undo and Go buttons, along with the familiar View button to access a screen where scenes can be saved, listed and recalled. Other functions available here include selecting which channels and parameters are 'safe' (ie. unchanged) during scene recalls, and what MIDI data is output during a scene change.
A set of freely assignable controls is arranged immediately below the Scene section, with four rotary encoders, four LCD panels, and eight push-buttons. A further trio of smaller buttons allows three separate sets of pre-configured, assignable controls to be recalled to the physical knobs and buttons. The ubiquitous View button is on hand to access another dedicated screen where any console function can be assigned to any knob or button. This facility makes it very easy to maintain instant access to critical parameters, such as a specific reverb send level, echo repeat time, and so on. The 'jump to page' function can provide handy instant access to a chosen display page, too.
Some other very useful functions are tucked away in the Setup display menu, the most important being one that ensures all 'scratchpad' configuration data is saved, enabling a 'safe' shutdown. Another option ensures that the main outputs are muted when the console is powered up or down, to avoid nasty noises! Normally the desk boots up to the last-used configuration (assuming the scratchpad data was saved correctly), but if the Scenes Undo button is held while the console is booted, it will restore the factory default settings instead. Pressing Undo again will reload the previous setup configuration. There are also options to initialise all console settings (without affecting stored Scenes), and to erase all Scene data. Finally, there is a handy function to lock out the control surface entirely!
We will be looking at this console again in the months to come after testing it more fully 'on the road', but my initial impressions are almost entirely favourable. It looks and feels very solidly constructed, the audio performance is impressive, the functionality is extremely comprehensive, and the few operational foibles are minor and easily overcome. Navigating around the desk is, by and large, very intuitive and logical, with only a couple of less-than-obvious features, like the input-channel naming function. The console layout is simple and clear — just what's needed for a live-sound situation — and the Scene memory facilities are comprehensive and powerful. I'm not sure how visible the screen or scribble-strip LCDs would be outdoors, but it's nice that they can be dimmed for low-light environments.
Usefully, the headphone outputs are capable of going extremely loud with low-impedance headphones, so you stand a chance of being able to hear what you're doing when soloing channels during a gig! The expansive and flexible I/O arrangements are more than sufficient to meet the needs of any typical installation, and the integration with the S16 stage boxes is a very attractive and cost-effective feature. The preamps are clean and usefully quiet, with sufficient gain for live-sound applications. The relatively high input impedance works very well with dynamic mics, and the ability to accommodate line-level sources is very useful. The channel EQ is flexible and sounds good, while the output EQ is even more versatile! The dynamics all work well too, with minimal artifacts, although the lack of an auto-release mode is disappointing. The built-in effects cover far more ground than usual, and achieve a far higher sound quality than would be expected of a console at several times the price of the X32.
Quality control has been Behringer's Achilles' Heel in the past, and only time will tell how this console fares in those departments, but initial reports from beta testers, along with my own experiences, are very encouraging, and the inclusion of a three-year warranty indicates the company's belief in the product. That being the case, this console really could turn the semi-pro and small professional digital live-sound market on its head as far as the accepted feature-to-cost ratio is concerned.
I can see this desk becoming very popular indeed in theatres and houses of worship, as well as with live-sound installations. It could also be employed as a studio console, although perhaps it isn't ideally optimised for that role quite yet. Nevertheless, the X32 is so well designed, built and priced that I'd say it's pretty much a no-brainer for a good many applications. The X32 should win the 'Best Product of 2012' award with considerable ease!
At the launch price, this console has no alternative — anything remotely comparable costs between two and four times more. Moreover, the X32 compares favourably from a features point of view with some very serious high-end live-sound consoles.
I performed a full set of test measurements on the console using an Audio Precision system, the results of which closely matched Behringer's published specifications and are very impressive. You can see the results at /sos/aug12/articles/behringer-x32-test-results.htm.
Free remote-control software called XRemote for Windows (with a Mac version to follow) will shortly be available on the Behringer web site, and a free iOS app to control the X32 from an iPad is also imminent. XRemote controls the console via its Ethernet connection, while a wireless router connected to the same port would be required for the iPad app. The iPad app allows the X32's monitor or main mixes to be adjusted in real time from the stage or auditorium via Wi-Fi, while the XRemote software provides complete control of all functions. Both of these systems were demonstrated in prototype form at the MusikMesse earlier this year.
The console operates with base sample rates of 44.1kHz or 48kHz, but the main reason for working at 48kHz is for compatibility with video systems, in which case clock synchronisation is an absolute necessity. However, there is currently no provision for word-clock in or out, although external clocking could easily be added with an interface card option, and hopefully that will be addressed soon.
The console's 'analogue in to analogue out latency' measured 0.85ms — a delay equivalent to standing nine inches further away from the source! Adding in the inherent 0.25ms additional network latency of the AES50 interface, analogue-to-analogue via a stage box would be almost exactly 1ms. This is an extremely good figure for a digital console and few can match, let alone better, that latency.
One of the few areas where such latency could be an issue is when using in-ear monitors (IEMs) for foldback, since performers hear their voice via direct bone conduction before the IEM signal arrives through their earphones, potentially creating a disturbing comb-filter effect. Different people can tolerate different amounts of latency, but 1ms is not usually too problematic for most.
A common way of reducing latency is to operate digital consoles at 96kHz, since that significantly reduces the analogue-to-analogue time through the desk, by halving the converter latency. Higher sample rates also often appeal to project studios. However, with a fixed amount of DSP available in the console, adding such an option would probably halve the number of available input channels and outputs, and I doubt that this compromise would make a 96kHz option very attractive.