Whether the project is live or installed sound, whether it calls for a customized solution or an off-the-shelf option that provides ease of use and no learning curve, there are a number of consoles and processors that fit the bill.
In January, the Allen & Heath SQ-Rack joined the SQ series. It combines SQ’s 96kHz XCVI core with connectivity and flexible control options in a compact, shallow 4U rack-mount form factor with multiple mounting options. It delivers 48 input channels with local I/O alongside an intelligent SLink port, integrating with Allen & Heath’s Everything I/O ecosystem. Further expansion is available with option cards for Dante, Waves, MADI, or additional SLink capabilities. Multichannel USB support enables easy recording and playback, whether connecting to a computer or directly to an external drive. It has a dedicated fader screen, customizable SQControl display, and tactile hands-on controls, plus SQ-MixPad, SQ-4You, and SQ-Control apps.
The Allen & Heath AHM Audio Matrix Processors suit fixed installation projects for multi-room audio distribution. The original AHM-64 has been used at the heart of many installations, from churches to large global media networks, and with the recent introduction of the smaller footprint AHM32 and AHM-16 matrix processors, integrators can quickly create systems that build from a single boardroom to a complete campus. Supported by scalable I/O, control, and Dante solutions, optional Acoustic Echo Cancellation (AEC) module, and including 8-band PEQ, gate, compressor, and delay on all input channels, and 8-band PEQ, 30-band GEQ, compressor, ANC, source selector, limiter, and delay on all zone outputs.
The ATEN AD004E converts Dante inputs into 4-channel analog line outputs, integrating analog and digital audio devices into one system through Dante AoIP media networking technology. It supports 24-bit and offers sampling rate options of 44.1k, 48k, 88.2k, and 96 kHz to meet different application needs. The built-in DSP allows 24 presets and speaker functionality management through the PC-based app Audio Wizard / ATEN VK control system via LAN / RS-232. A DSP matrix mixer is also contained to deliver DSP mixing and routing functions, enabling users to assign Dante inputs to any of the four analog outputs. The AD004E is AES67 compliant, elevating interoperability among different audio networking protocols and devices. Plus, the unit comes with a selectable automatic Standby Mode when the signal level runs lower than -50 / -60 / -70 dBu for a selected duration (10 / 30 / 60 minutes). The AD004E supports redundancy operation thanks to its PoE functionality, eliminating the need for additional cable installations. All the features come in a compact half-rack size enclosure, and consequently, less rack space is needed, and installation under desks or on any flat surface is made easy. Not only does the AD004E enhance interoperability among AoIP devices and allow for speaker management; it also realizes distributed DSP systems when working with ATEN Pro Audio models, elevating installation flexibility, reducing overall cost, and broadening applications. It is suitable for professional audio scenarios utilizing the Dante networking technology, such as boardrooms, conference rooms, and hospitality venues such as restaurants and bars.
Released last May, the inexpensive Audio-Technica AT-UMX3 is AudioTechnica’s first USB audio mixer for streaming and other applications. Drawing upon AudioTechnica’s pro audio heritage, the mixer was designed and tested by top A-T engineers. It was built to be easy to operate even for firsttime users. This all-in-one interface is equipped with one XLR / 1/4-inch microphone input, two LINE inputs for guitar and keyboard, and a headphone jack for monitoring. The AT-UMX3’s loopback function allows users to mix mic and instrument audio with sound from their computer into a single stream. When loopback is turned on, it is possible to play background music while streaming or to stream game sounds and voices at the same time. Features include a microphone mute function to quickly silence the microphone, and a microphone monitor mute function to mute the mic audio only in the mix sent to connected headphones. The mixer’s high-performance A/D converter provides resolution up to 24-bit/192 kHz, ensuring crystalclear output for voice and instruments. Additionally, the AT-UMX3 features a noise-resistant design to prevent the pickup of unwanted electronic noise from smartphones, Wi-Fi routers, and other sources. It’s equipped with a Neutrik XLR/6.3 mm (1/4-inch) combo microphone jack, stable 48V phantom power supply, a Hi-Z input jack for direct connection of an electric guitar, and a stereo input for a keyboard. The system is compatible with Windows and macOS, iOS, iPadOS, and Android OS. Compatibility has also been verified with major live-streaming apps and online chat apps. The mixer comes with a USB cable (USB Type-C/USB Type-A) and a USB conversion adapter (USB Type-A/USB Type-C).
The Audio-Technica ATDM0604a Digital SmartMixer is an update of the popular ATDM0604. The ATDM-0604a offers advanced functionality for use in applications from meeting spaces to educational facilities. Incorporating user feedback from leading system integrators, the enhanced functions bring improved performance for online or live meetings. Improvements include enhanced echo and noise-cancellation performance; LED-based remote control of the Audio-Technica ES954 hanging microphone array via the GPO terminal; cascade connectivity for up to eight mixers sharing audio buses and SmartMix controls; and mic/line level support for input channels 1 to 6. The SmartMixer technology allows channels to be mixed automatically in gate or gain-sharing mode, ensuring consistent output from all inputs in a setup. In addition to a Web Remote interface, controls and LED indicators on the mixer’s front panel support: adjustment for input/output and gain levels; set and recall presets; on/off for phantom power, low-cut filter, automatic mixing and acoustic echo cancellation (AEC) by channel; IP configuration changes (Auto or Static); noise cancellation assignments, and more. DSP is available for inputs and outputs. Input processing includes a four-band parametric EQ, compressor/de-esser. Eight 8-band feedback suppressors can be assigned to an input or output. Output processing includes a 12-band parametric EQ, compressor, and limiter. IP-based external remote control are supported.
In February, the new DiGiCo Quantum225T debuted, reconfigured to provide specialist programming tools for theatre sound design, rehearsal, and show operation. In particular, Quantum225T’s programming and workflow enhance the cue system with DiGiCo’s Auto Update and cue data management tools. Auto Update allows designers to establish intricate inter-cue relationships, with changes made to channel settings automatically propagating to other related cues. Character variations, often a result of costume and prop changes, are handled with the Alias function, and cast changes are easily managed through the Players function. Quantum225T’s channel processing and mixing functions are identical to its live-focused Quantum225 counterpart, but the crosspoint matrix gains individual nodal delays and matrix aliases. It carries features from the existing Quantum Range, including Mustard Processing channel strips, Spice Rack plugin style native FPGA processing options, Nodal Processing, and True Solo. The Quantum225T is equipped with 96 input channels and 48 busses, as well Mustard and Nodal Processor tallies to 36 and 48, respectively. Mix minus is also included in the new feature set, plus a larger 24×24 matrix. There are four MADI ports and dual DMI ports for added connectivity, 8×8 analog and four AES channels for local I/O, built-in UB MADI, and optional Optocore, plus dual PSU.
In March, d&b audiotechnik announced new scalable I/O sizes for the d&b DS100 and DS100M signal engine, an extension to the d&b Soundscape immersive ecosystem. With the introduction of scalable I/Os, the DS100 firmware has been updated to offer three license sizes, allowing customers to select their preferred I/O count, whether to set a cost-effective entry point or benefit from increased channel capacity. Existing DS100 and DS100M users receive a free firmware upgrade, increasing their maximum I/O processing from 64×64 to 128×64. Additionally, two new DS100 I/O sizes providing 64×24 and 64×64 were introduced, offering a broader range of price points and application sizes. The licenses are upgradeable at any time, allowing users to tailor I/O to their specific projects and needs without the need for additional hardware. With the introduction of the scalable solutions, d&b Soundscape can now support smaller applications such as houses of worship, clubs, restaurants, and mobile setups, while simultaneously enhancing complex productions like musical theatre, where higher input counts simplify programming and scene management. In addition to the I/O licenses, the available function groups have also been doubled to make more complex systems easier to manage. En-Scene will now come pre-installed on all future DS100, ensuring that users can immediately benefit from its advanced object-based mixing capabilities.
The Electro-Voice N8000 NetMax 300 MIPS digital matrix controller delivers full IRIS-Net supervision, control, and scheduling along with up to 32 input channels, extensive DSP, and CobraNet audio and control connections. It offers support for Ethernet, RS-232, USB, and CAN, and Dante audio networking options are available. Four slots with 8-channel audio modules form the foundation for flexible customization, and each slot can house input or output cards. It includes internal 48-bit processing, and the auto-compiling DSP engine has ultra-low fixed latency. One time or regularly scheduled events can be arranged and reactions to certain events or system states can be configured. Any system problems can be detected automatically and can be displayed on the PC screen or transmitted to external sites.
The Extron DMP 128 Plus and DMP 64 Plus audio processors are certified for Microsoft Teams. The DMP 128 Plus AT models support Dante Domain Manager and compatibility with the VoIP MS cloud-based VoIP system for the DMP 128 Plus V models. Extron DMP processors are also compatible with the Avaya XI workplace solutions. Extron is committed to broad compatibility for these processors.
Meyer Sound Galileo GALAXY is Milancertified and utilizes open-source AVB technology to drive and align loudspeaker systems with multiple zones. It provides a powerful toolset for corrective equalization and creative fine-tuning in applications from touring to cinema. Building on Galileo’s algorithms, GALAXY retains users’ favorite processing tools, including the patented five-band U-Shaping and parametric EQ, while adding a new crosspoint delay matrix feature and improved delay integration. Three versions are available for different configuration needs. The newest generation of FPGA-based processing with up to 64-bit resolution delivers increased dynamic range, a lower noise floor, and super-low latency of 0.6ms analog into analog out.
Meyer Sound NADIA is an integrated, network-based digital audio processing and distribution platform that powers Constellation by Meyer Sound. When incorporated into new Constellation installations, in addition to the processing power and inputs reserved for active acoustics, NADIA also provides separate inputs, processing, and matrixing to enable Spacemap Go spatial sound with no performance compromises for either function. The NADIA platform supports up to 96 inputs for Constellation acoustic processing as well as 128 independent program audio inputs and comprises three hardware modules. All NADIA-based systems require at least one NADIA-CP core processor that supports 128 outputs. Additional NADIA-CP modules can be added. All communication to and from the NADIA-CP module is via a Milan-compatible AVB network. All processing is at 96 kHz/64-bit floating point resolution. The NADIA-AI12 input module provides 12 channels of analog input with a preamp on each channel. The NADIA-AO16 provides 16 channels of analog line-level output. NADIA outputs can be routed directly via the network to Milan endpoint loudspeakers such as ULTRA-X20 series compact loudspeakers and USW-112P subwoofers. Each NADIA-CP module hosts up to 12 VRAS (Variable Room Acoustic System) processors, enabling the configuration of up to 12 discrete acoustical zones in a single unit. For scalability and cost efficiency, two licenses are available: standard for all 12 VRAS processors, and lite for three processors at a substantially lower licensing cost. When compared to the prior D-Mitri processor solution, Constellation systems based on NADIA will benefit from reductions in both rack space and overall costs.
The L-Acoustics L-ISA Processor II is a second-generation hardware solution that provides advanced object-based mixing for immersive audio productions. Like its predecessor, it supports spatial audio processing and virtual acoustics for up to 96 audio objects based on speaker positioning information and mixing parameters—including pan, width, distance, elevation, and aux send—the new L-ISA Processor II doubles the original unit’s potential output count of 64 up to 128 outputs for greater power and versatility on larger, more complex events. Seeing that most productions are unlikely to utilize 100-plus outputs, L-Acoustics is offering L-ISA Processor II in a choice of four output counts—16, 32, 64 or 128—from the same device, with various capacities accessible via different lifetime licenses at tiered pricing levels. For example, a small club or theater may need no more than 16 outputs, while a mid-sized house of worship or performing arts center might require as many as 32. With L-ISA Processor II, those customers now have access to all of the very same immersive tools, 128 inputs and premium quality 96 kHz sampling as large-scale musicals or massive tours running triple-digit outputs, but at a scalable cost.
Lectrosonics DNT Series processors operate as native DanteTM devices to integrate analog and digital network signals. Analog audio signals are converted to digital signals that are then processed and assigned to Dante network channels. Digital signals from the Dante network are processed in the same manner as analog signals, then sent back into the Dante network or converted to analog for mic/line outputs. The result is an integration of analog and digital audio signals into a common network where any signal assigned to a Dante transmit channel is then available at any receiving device connected to the network.
At NAMM, the Mackie ProFX10 GO mixer debuted. Providing up to eight hours of power out of rechargeable, swappable batteries, the 10-channel analog mixer is encased in a steel chassis and uses Mackie’s Onyx preamps and offers built-in GigFX+ digital effects, EQ and compression, as well as USB-C interface and Bluetooth connectivity. The ProFX10 GO’s Onyx preamps provide up to 60 dB of gain and 48V phantom power on all channels, along with analog compression and a 3-band EQ. The GigFX digital effects engine can be used to edit and save more than 24 different reverbs, delays, and choruses. Bluetooth connectivity allows users to send and receive audio wirelessly from iPads, phones, and other devices, aiding a variety of uses, such as playing DJ set, performing to backing tracks, or livestreaming. Using the USB-C connection, users can record to a computer at highresolution sample rates up to 192 kHz.
The Mackie M•Caster Studio is the flagship of the M•Caster Series of live streaming mixers, designed to be the heart and hub for Twitch-based gamers, YouTube Live creators, podcasters, and live streaming professionals. Features include a built-in sampler and full color MixViewer display. For Twitch-based, fast-paced content creation, M•Caster Studio plugs in directly or connects via Bluetooth and offers integrated ContourFX presets and voice manipulation via the StreamFX feature
The Shure IntelliMix P300 conferencing mixer connects up to 10 Dante audio inputs, two analog inputs, a USB soft codec, and a mobile device. Each input channel can be auto-mixed and provides acoustic echo cancellation, noise reduction, automatic gain control, matrix mixing, delay, compressor, and PEQ. An additional 3.5mm connection enables attendees to join by tablet or phone, and all this fits into a half-rack space package. It connects to room systems through two input and two output terminal blocks while its ethernet connection can use PoE to eliminate a separate power supply. Additionally, a predefined matrix and presets can simplify the setup process. Zoom and Teams certified.
The SSL Live V6 software update unveiled at NAMM for the flagship SSL Live L650 brings acclaimed studio tools to the live stage. The Fusion effect rack emulates five circuits from SSL’s award-winning Fusion hardware, delivering rich tonal color. The Path Compressor Mix Control introduces advanced parallel compression directly to channels and buses, while updates to the TaCo app allow engineers to remotely control the SSL Sourcerer and Blitzer modules. Further, enhanced Dante Routing Modes provide seamless system-wide integration across ShowFile Saved and Outside of ShowFile setups. Built on an open, flexible architecture, SSL Live allows operators to configure workflows for any application. Advanced routing capabilities make it equally suited to touring setups—supporting up to eight SuperAnalogue MADI stageboxes via the Blacklight II Concentrator interface—or installed sound systems with full Dante routing for multi-room configurations. All this versatility is now accessible through SSL Live Bundles, a cost-effective solution providing complete systems optimized for touring, installed sound, and church audio.
The Peavey PCX Series of digital matrix speaker management processors includes three units of processing technology with I/O outputs that can be routed in multiple configurations to meet virtually any requirement. They perform loudspeaker management functions as well as matrix mixing, room combining, and other audio processing functions; the PCX Series is well-suited for installed and portable sound systems. The PCX 260 offers 2 XLR inputs and 6 XLR outputs; the PCX 480 offers 4/8 XLR I/Os; and the PCX 88 offers 8/8. Notably, on the PCX 88, an additional 8 digital input channels and 8 digital output channels, all independently assigned, are available over Ethernet with the optional Dante network audio card. On each PCX Series processor, the crossover filters are fully adjustable, and any input can be routed to any output via the digital matrix system, allowing for any speaker management cabinet configuration.
The Soundcraft by HARMAN Vi1000 digital mixing console is a 34in. x 32in. complement to its two larger siblings, the Vi2000 and 3000. The Vi1000 employs Soundcraft’s Vistonics II channel strip user interface, as well as SpiderCore, a built-in DSP and I/O engine based on STUDER technology, offering 40-bit floating point processing. The Vi1000 features an additional remote-control surface for any of the larger consoles in the range using their Mirroring feature. The Vi1000 comes with rear panel local I/O featuring 16 HQ mic/line inputs, 16-line outputs, and two 64-channel expansion slots that allow up to two MADI-based Stageboxes to be connected, or alternatively, the slots provide access to an extensive range of D21m I/O option cards that address all industry-standard audio formats. Four channels of AES/EBU I/O plus USB and MIDI connections complete the back panel. The total I/O count of the console is 212 in x 212 out.
The Presonus StudioLive 64S brings the power of a large production console to small-format digital mixing, with 76 mixing channels, 43 buses, and 526 simultaneous effects thanks to the quad-core FLEX DSP engine. StudioLive 64S mixers feature an independent main Mono/ Center bus; each channel has a dedicated level control and a Center Divergence control that allows users to control the pan placement in their LCR mix. Powered by the PreSonus FLEX DSP Engine, it features state-space modeled Fat Channel plug-in processing on every input and bus; up to 32 FlexMix buses that can be individually configured as Aux, Matrix, or Subgroup buses; and flexible digital patching. It delivers 128-channel (64×64) USB re cording, plus multitrack SD recording and extensive 128-channel AVB networking.
The TASCAM Model 2400 live recorder and mixing console with audio interface builds the attributes found in the company’s Model 12, Model 16, and Model 24. The new flagship Model 2400 provides an integrated 24-track digital recorder, a 22-channel mixer, and a 24 In / 22 Out USB audio interface. It’s an all-in-one choice for use in live recording / rehearsal environments and related sound reinforcement applications. With its 24-track digital recorder that can capture the recording and mixdown of live performances to an SD, SDHC, or SDXC card without the need for a PC or other DAW (Digital Audio Workstation), the Model 2400 is a self-contained workstation that offers tremendous functionality in a compact form factor. With its integrated USB audio interface, the Model 2400 facilitates linked operation with DAW control for extensive editing and high-quality production. Equally notable, the Model 2400 is easy to use, with a bus configuration that provides excellent operability characteristic of an analog mixer, with a design that facilitates compatibility and expandability with other studio peripherals via its MIDI interface. In addition to the aforementioned features, the Model 2400 also offers 16 premium TASCAM XLR mic preamplifiers, 12 channel inserts, 5 Aux sends, and 4 stereo sub mixes. This versatile workstation’s MIDI In /Out ports include support for MIDI Time Code (MTC) and MIDI Clock/ Song Position Pointer (SPP) Out, making it easy to synchronize, for example, an entire MIDI keyboard/ sampler setup as part of the overall mix components. There is also a Click Out jack with a TAP Tempo function.
The Waves Cloud MX Audio Mixer is a 100% cloud-based audio mixer, for audio, mix control, and creative processing with Waves plug-in integration. Cloud MX remote operator control options include support for touch screens, plus tactile mixing with the Waves FIT Controller. The Waves Cloud MX Audio Mixer is fully Dante Connect compatible for up to 256 channels of remote audio capture and full cloud production.
The Xilica Solaro QR1 is a quarter-rack Linux-based DSP processor that has eight modular card slots that accept 2-channel audio input and output cards and 4-channel GPIO cards selectable as input or output. The ability to use any card type, in any combination, in any card slot maximizes I/O flexibility and allows designers to specifically customize I/O without waste. A 4×4 Dante card is built in. Supported by Xilica’s drag-and-drop Designer, it is compatible with various Xilica control interfaces as well as any third-party control system. External power supply included; also PoE. Optional AEC mic inputs up to 8 channels@250ms and 16 channels@100ms.
The Yamaha DM3 Series includes a pair of 22-channel ultra-compact digital mixing consoles—the DM3-D and DM3 Standard. The DM3-D has the added feature of supporting Audinate’s Dante protocol, allowing it to tie into a Dante audio network, but the desks are otherwise identical, providing users with 16 mono, one stereo and two FX Return input mixing channels, as well as a half-dozen Mix Send, two FX Send, one stereo and two matrix output mixing channels. Also onboard are 18 effects; a full-color 9-inch touchscreen; a dark mode, and some features intended to make it useful outside of live sound— namely, enabling it to be used in recording situations as a DAW remote control.
