A new breed of point source speakers from d&b audiotechnik

The demands for a high performance point source have always been the same: reliable, lightweight and compact, and with a wide range of rigging options for maximum flexibility. The V7P and V10P from d&b audiotechnik are a new breed of speaker offering all this and more, including the d&b hallmark constant directivity control down to low frequencies.

d&b dévoile les V7P et V10P

d&b dévoile les V7P et V10P


Building on the 3-way passive design of the well known d&b V-Series line array modules, the V7P and V10P point sources house two 10” drivers in a dipolar arrangement with a rear mounted 8” MF driver attached to a dual chamber horn. The exits from this horn design create another dipole around the centrally mounted single 1.4” exit compression driver with a constant directivity horn. This maximises the performance to size ratio, by making almost the entire baffle board radiate.

d&b dévoile les V7P et V10P

The V7P and V10P deliver 75° x 40° and 110° x 40° (h x v) dispersion characteristics respectively. All components are arranged symmetrically around the centre axis of the cabinet to produce a perfectly symmetrical dispersion pattern.

Due to the dipolar arrangement of the LF drivers, a broadband, horizontal dispersion control is maintained down to approximately 350 Hz. The rotatable horn and wide range of rigging options make mounting in either orientation simple.

The bass-reflex design of the V7P and V10P utilizes an advanced LF venting arrangement that delivers an impressive extended LF performance. The frequency response extends from 59 Hz to 18 kHz offering flexibility as a stand alone solution.

d&b V-GSUB

The new V-GSUB is the ideal companion to the V7P and V10P for ground stacked applications.
This high performance cardioid subwoofer requires only one amplifier channel and shares the same acoustical performance, cabinet design and driver arrangement as the V-SUB :
An 18” driver in a bass reflex design facing towards the front and a 12” driver in a two chamber bandpass system radiating towards the rear.

The cardioid dispersion pattern resulting from this driver arrangement rejects energy towards the rear and reduces the excitation of the reverberant field at low frequencies. The V-GSUB is fitted with runners and stacking recesses to prevent unwanted movement when stacked with V7P or V10P cabinets.

The Vi7P, Vi10P and Vi-GSUB are installation versions of these versatile new speakers, offering visually discreet cabinet design and installation specific rigging systems. Special Colour (SC) and Weather Resistant (WR) finishes ensure they fit seamlessly in any permanently integrated application.

Product Manager Werner ‘Vier’ Bayer commented : “These point source speakers are an ideal alternative to smaller line array setups, especially considering their lightweight design and wide range of rigging options. The V7P and V10P use an advanced hornloaded mid range section, which provides a remarkable MF sensitivity.
This exceptional MF performance, particularly in the vocal range, is where these compact V-Series cabinets excel. This endless vocal headroom makes them perfect for applications which demand high sound pressure levels with full bandwidth capabilities from a single box solution. ”

More information : www.dbaudio.com/

News from Prolight + Sound

Yamaha TF Series, digital consoles for small budgets

Yamaha continues to expand its range of digital consoles to accommodate lower budgets, with the introduction of the TF Series, which follows the CL and the QL Series. Also exhibited on the stand of the Japanese manufacturer was the PM10 Rivage digital mixing system – the flagship of the brand (see SLU) – and RSiO64-D, a rack of mini-YGDAI cards for inputs and outputs that accepts more than 30 types of cards .

TF Series digital consoles

Yamaha TF5

The TF range includes three models of compact digital mixing consoles: TF5, TF3 and TF1, which offer 33, 25 and 17 motorized faders respectively and 32, 24 and 16 analog inputs on the rear panel.
All three are equipped with the new Yamaha D-PRE mic preamps with storable settings, a feature that is particularly appreciated in live performance, where it is necessary to make configuration changes quickly. These consoles operate with a sampling frequency of 48 kHz.

The smallest of the TF Series consoles, TF1, features a rack-mountable, 19″ format chassis

The TF Series offers the user a total of 48 mixing channels on TF3 and TF5, and 40 on the TF1, counting the analog inputs, the USB stereo digital inputs and the two effects returns.

As far as recording is concerned, TF Series consoles can record and play back up to 34 channels via USB 2.0 to a PC or MAC (using Nuendo Live), and two channels using USB media.

The TF Series consoles are equipped with a slot for the D-NY64 I/O expansion card, which allows them to send and receive up to 128 channels (64 in/64 out) within a Dante audio network.

Using the TF5 console with this card installed together with the new Yamaha I/O rack, Tio1608-D, it is possible to create a console/stage box system with up to 48 inputs and 24 outputs, counting the sixteen outputs on the console.

The rear panel of TF5, with its 32 mic/line inputs and sixteen outputs.Two stereo line inputs are also included on all TF consoles. Note the covered slot for the NY64-D card, that will be made available later.

The color LCD touch screen permits multi-touch control of settings such as parametric EQ curves. The 1-Knob Com (for dynamics) and 1-Knob EQ (for equalization) functions make adjustments easy by allowing the user to touch only one encoder for control. Finally, another feature makes its debut, the “GainFinder”, which makes it easier to achieve the optimum setting of the input gain.

The TF Series consoles include the “One-Knob” feature that allows adjustment of the EQ or compressor with a single encoder.

 

The touch screen responds “to the fingers and to the eye”


The TF Series also has powerful processing and effects. In addition to the parametric equalizer and the two dynamics processors on each channel, the console offers eight processors, two of which are globally accessible and offer the equivalent of industry-standard Yamaha SPX processors. The remaining six processors are available on the Aux buses 9/10 through 19/20. The Aux buses 1-8 and mains offer the same 4-band parametric EQ as the input channels, complete with a 31-band Flex12 graphic equalizer.

For less experienced users, Yamaha has included preset configurations (equalization, dynamic, gain, etc.) for the most common Audio-Technica, Sennheiser and Shure microphones, which have been developed in collaboration with these manufacturers and with renowned sound engineers (QuickPro presets). These offer good base settings for use with musical instruments and voices. The same system has been adopted for the outputs, with presets for in-ear monitors and for Yamaha powered speaker systems.

Finally, separate apps are available for wireless control of the mixing, for personal monitor mixes and for offline setup: TF StageMix for iPad, MonitorMix for iPhone, iPad or iPod touch, and TF Editor for Windows or Mac computers, respectively.

The RSio64-D I/O rack

The Dante/Mini-YGDAI interface rack, RSio64-D

RSio64-D is a Dante interface rack for Mini-YGDAI cards, which offers 64 inputs and 64 outputs. It is remotely controlled by the CL or QL Series consoles or via the R Remote software. Equipped with four slots, it can host more than 30 different models of Yamaha Mini-YGDAI cards. It allows the integration into a Dante network of a wide variety of input/output formats, such as AES/EBU or ADAT. It is not limited only to I/O cards, as it also accepts DSP cards, such as Lake processing or Dan Dugan automatic mixing.

Each of the Mini-YGDAI slots is equipped with a sample rate converter, which allows the connection of devices referenced to different clocks. Also, Rsio64-D supports Dante network redundancy (it includes primary and secondary ports), and has an input for external powering to guarantee power supply redundancy.

To facilitate its use as a format converter and router, presets are provided for seven basic routing configurations, including routing between the Mini-YGDAI cards. A rotary switch on the front panel allows direct selection of Dante/Mini-YGDAI routing, crossover routing from Mini-YGDAI to Mini-YGDAI or other options designed for a wide range of applications in broadcast or sound reinforcement.

More info on the Yamaha site: http://www.yamahaproaudio.com/europe/fr/products/mixers/tf/index.jsp

 

 

 

 

News from Prolight + Sound

RCF HDL50-A, a new amplified system

Numerous new products were presented by RCF at this edition of Prolight + Sound: a new 3-way, amplified line array system, HDL 50-A; an addition to the TT series, the TTL6-A amplified enclosure; the new stage monitors ST 12-SMA and ST 15-SMA; and finally, a series of small rack-mount digital mixing systems including the M08 and M18.

Here, you can clearly see the completely symmetric configuration of the HDL 50-A enclosure

The newest and the largest line array of the D Line series is a three-way system with integrated signal processing and amplification. Its cabinet, with a mixed construction of birch plywood and molded polypropylene, combines with the use of transducers with neodymium magnets to give it an extremely low weight (48 kg), considering its onboard amplification and the sound pressure levels it can deliver.

RCF HDL50 ligne

 

 

 

A single HDL 50-A module incorporates two 12″ long-excursion woofers, with neodymium magnets, in a coplanar configuration framing four 6″ midrange speakers, and two compression drivers with 3″ diaphragms coupled to an RCF “4 Path” waveguide. This waveguide, which provides a dispersion of 90° H x 10° V, creates an isophasic wavefront from 700 Hz up.
The use of the two ND850 compression drivers with 3″ voice-coils has allowed the designers to set the mid-high crossover point quite low, 800 Hz, covering most of the vocal range.

The 4400 watts of on-board class D amplification is divided into three modules: 800 W (HF), 1400 W for the four midrange drivers and 2200 W for the two 12″ woofers. It is preceded by signal processing via the 48 kHz/32 bit floating-point DSP. Monitoring and system configuration is carried out via the proprietary RCF network, RDNet. A single HDL 50-A module can deliver a maximum of 140 dB SPL.


Gioia Molinari (Director of Marketing) presents the new TTL6-A enclosure

The TTL6-A is also a three-way active speaker that can be mounted in a column of two units using a purpose-built rigging system. It utilizes components similar to HDL 50-A, namely two 12″ neodymium woofers for the low end and four 6″ neodymium transducers for the lower midrange. These surround a compression driver with a 2″ throat (3″ v.c.) loaded by an asymmetric waveguide with a vertical dispersion of 30° (+5°/-25°). With its horizontal dispersion of 90°, this enclosure will serve in both touring and installation applications.

The open TTL6-A provides a glimpse of the asymmetric HF waveguide

The onboard amplification/processing section uses four class D amplifier modules: 2 x 550 W for the bass transducers, 700 W for midrange, and 400 W for the high frequencies.

In addition to the connectors, the rear panel provides access to settings via an encoder and a display. But, as with HDL 50-A, control, monitoring and configuration can be carried out via RDNet (ver. 2.2). The DSP takes care of the three-way crossover, limiting and compression, protection and the temporal alignment of the transducers.

After opening the foray into the world of consoles last year with the L-Pad series of compact mixers, this year RCF sets out into the field of digital mixing with the M08 and M18 rack mixers without control surfaces. The role of control surface is covered by a touch-screen tablet running an application. These consoles are designed more specifically for musicians.

The digital rack mixer M18

M18 offers eight mic/line inputs with discrete preamps and digital gain control, while the M08 has four. Two of the inputs can be switched to high impedance to receive signals from guitars or basses. Ten additional line inputs are included in the M18, four in the case of M08. As far as outputs are concerned, M18 offers the stereo master plus six auxes, while the M08 is limited to two aux outputs.

Both consoles have dual WIFI connection at 2.4 and at 5 GHz and can utilize external antennas to improve the link in difficult RF environments. Each audio channel – inputs and outputs – includes a four band parametric EQ with a choice between standard, vintage and “smooth” types, a compressor/limiter and gate. Both models include several types of effects (reverb, delay, chorus, flanger etc.) and an integrated stereo recorder/player (.wav, .mp3 and .aiff) via USB. Finally, the high impedance inputs can also add amplifier-modeling effects as inserts.

 

 

News from Prolight + Sound 2015

d&b Audiotechnik unveils D20 and MAX2

We discovered three big news on the d&b stand: the D20 amplifier, little brother of the D80, the stage monitor MAX2, successor of the German manufacturer’s renown MAX and, on the simulation-prediction software side, version 8 of ArrayCalc, which now has the ArrayProcessing extension.

The D20 looks just like its big brother, D80, from which it also inherits its on-board signal processing, its LCD color touch screen user interface, and the same two-rack-unit chassis, as well as the configuration and monitoring software R1 V2. It can be configured via Ethernet through the OCA protocol (Open Control Architecture). Just like its predecessor, it includes a web server, allowing the use of the interface and functionality of the LCD screen on a PC or MAC using your favorite browser.

The D20 with its LCD color touch screen

It differs from the D80 in that the available power on the four channels of class D amplification is 4 X 1600 W into 4 ohms, but the D20 still has a universal switching power supply with power factor correction. This allows it to overcome, to a great extent, variations in mains voltage and allows it to offer a very efficient ratio of power consumption to power output.

Within the d&b range, the D20 should serve as the amplification platform for small and medium systems, namely the E and T series point-source enclosures and the Y and V-Series line arrays. For the J Series systems and M2 stage monitor, the D80 is still required. Therefore, this new amplifier is basically a replacement for the D12.

The MAX2, worthy successor to the MAX, employs a 15″ coaxial transducer.

MAX2 is based on MAX, the wedge monitor widely used throughout the world, which it replaces. This coaxial stage monitor, constructed in marine-grade plywood, offers a wide frequency response from 55 Hz to 18 kHz, with a 75° conical dispersion. It incorporates a 15″ transducer with a ferrite magnet, which includes a coaxial 1.4″ compression driver. Both transducers share the same magnetic component.

This passive enclosure can also be pole-mounted on a sub and used as a small PA system. Four M10 inserts allow it to be mounted on a bracket. Specific presets are programmed in both the D20 and the D80, with which the MAX2 is capable of developing a maximum SPL of 135 dB. With its nominal 8 ohm impedance, the MAX2 has an applicable power rating of 1600 W peak (10 ms) and 250 W RMS.

d&b Audiotechnik ArrayProcessingArrayProcessing is an added feature in version 8 of the d&b ArrayCalc prediction software that can calculate, determine and improve the overall behavior of a line array system, both in terms of tonal balance over the spatial distribution of the audience and in terms of sound levels. ArrayProcessing functions by creating a set of FIR and IIR filters (and delays) for each module in an array, to correct the energy distribution and the frequency response in all the zones of the listening area.

Consequently, it takes as many channels of amplification as there are boxes within an array pre-set with ArrayCalc, although the boxes can have different horizontal coverage, for example J12s (120°) at the bottom of an array of J8s (80°). ArrayProcessing can only be used with the D20 and D80 amplifiers because the software requires the OCA control protocol on the amps. Due to filtering, this feature adds a latency of 5.9 ms to the 0.3 ms latency of the amps, for a total of 6.2 ms.

 

 

 

News from Prolight + Sound 2015

Nexo launches the ID Series multipurpose speakers

Nexo ID24iNexo chose Prolight & Sound for the debut of the ID24 series (ID stands for Inspace Definition), which comprises a line of ultra-compact, full-range speakers and two subs, S110 and S210.

Because there are numerous versions – between the touring and installation versions and the “à la carte” variants – the ID24 enclosures cover a wide range of applications, starting with under-balcony fills, front-fills for theatre or concert stages, or even monitoring applications.

An ID24t mounted on a truss structure with the Nexo QRV bracket. The installation can be done without tools.

ID24 incorporates two 4″, long-excursion, shielded transducers mounted in a ‘V’ configuration around a 1″ (½” exit) compression driver loaded by one or more specially-developed waveguides.

These very compact speakers (309 mm W x 132 mm H x 233 mm D) can deliver an impressive 126 dB peak SPL, thanks to their nominal sensitivity of 100 dB (1 W @ 1 m) over the band of 110 Hz–20 kHz (+/-3 dB) and their peak applicable power of 400 W.
Their nominal 16 ohm impedance allows the use of a single NXAMP4x1 Powered Controller, in 4X4 mode with dedicated presets, to drive 16 enclosures. If they are amplified using some other platform, the Nexo DTD controller supports the same presets.

The back of an ID24t enclosure. Note the waveguide rotator.

A key feature of ID24 enclosures is that they can be used both vertically and horizontally, with the possibility to rotate the HF waveguide using a rotating knob on the rear of cabinet. Therefore, these enclosures offer two different directivity patterns in each plane.

The basic versions – Touring, ID24t and Installation, ID24i – exhibit a directivity of 120° H x 30° V (or vice versa) and differ only in finish (grill and mounting hardware) and the type of connector: Speakon NL4 (touring) or captive-cable terminal block (installation).
The IP55-rated ID24i is available in white and black and can be used outdoors.

The ID24c “à la carte” version is available with three additional waveguide options: 60° x 60° and 90° x 45° symmetrical dispersion, and 60°-120° x 45° asymmetrical. It is also available in a wider choice of colors.

The cabinet of ID24 is constructed in a molded polyurethane composite that offers lightness, rigidity and strength. Numerous mounting points are provided on the metal plates of the top, bottom and sides. These allow the enclosures to be installed without using tools on Nexo (QRV bracket and QRF support) and third party mounting accessories, as well as pole mounts, truss mounts, wall mounts or on stands.

Two very compact and low-profile subwoofers complete the series : ID S110, with a single 10″ long-excursion woofer, and the dual-10″ ID S210. These subs feature cabinets in birch plywood and are loaded in an acoustic band-pass configuration.  Their low profile and small footprint allow them to be easily concealed in an installation.

The ID24t stacked on an S110 sub using the Nexo QRF support.

The S110 has a sensitivity of 97 dB (1 W @ 1 m) and can generate a peak SPL of 125 dB, while the S210 (sensitivity: 100 dB) can deliver 131 dB peak.
The crossover points for the full-range heads can be varied between 85 Hz and 120 Hz and the two subs offer a frequency response (-3 dB) that extends down to 45 Hz.

 

 

Symetrix Announces SymNet Composer 3.0

Symetrix has launched SymNet Composer 3.0, an update to the manufacturer’s award-winning open architecture design software. Version 3.0 has built upon the native support of Audinate’s Dante media networking technology for select third-party devices in earlier releases, and extends support of market-leading products thanks to new partnerships with Shure Inc. and Audio-Technica.

Symetrix Symnet V3

Now, integrators can streamline their set-up procedures and achieve network discovery, Dante signal routing, and audio set-up of supported third-party devices from the two new manufacturers alongside Attero Tech and Stewart Audio with a single piece of software.
For Shure users, SymNet Composer 3.0 brings the MXWAPT4 and MXWAPT8 access point transceivers from Shure’s Microflex™ Wireless range into the SymNet fold.

SymNet Composer 3.0 also adds native Dante configuration for two leading products from Audio-Technica. The two supported items are the ATND971 Dante-enabled cardioid condenser boundary microphone and the ATND8677 Dante-enabled microphone desk stand, which can be used with any gooseneck microphone sporting a three-pin XLRM-type output connector.

The new version of SymNet Composer also includes a number of Window Management (Composer UI) Framework improvements, giving the user greater freedom to move pallets to any side of the screen they desire, resize them in place and nest them into tabbed pallets. It is also possible to create specific menu bars for frequently accessed commands and customize toolbars to keep preferred shortcuts at hand.

SymNet Radius 12×8 EX
Alongside the introduction of SymNet Composer 3.0, Symetrix also launched the SymNet Radius 12×8 EX. An upgrade to the popular SymNet Radius 12×8 DSP, the new version features the addition of an expansion slot which, when utilized, increases the total audio input/output count from 20 to 24 in the same 1U format.

Symetrix Radiux EX back

In a development that boosts the future-proofing of an installation by allowing for subsequent expansion by simple I/O card addition, SymNet Radius 12×8 EX supports an array of optional SymNet cards, including analog, digital, AEC and telephone. A dedicated Migration Tool allows those with previous designs or work-in-progress using Radius 12×8 to easily convert their files to use the Radius 12×8 EX hardware without any further modifications or extra work.

SymNet 2 Line VoIP Interface Card
Also available is the 2 Line VoIP Interface Card, which is an Asterisk®- and Cisco®-compatible plug-in card for the SymNet Edge and Radius DSPs that natively integrates with SIP-based call platforms and unified communications environments. Designed to be easy to deploy and manage, the 2 Line VoIP Interface Card is suited for a variety of conferencing, paging, remote monitoring and broadcast applications.

Symetrix founder and CEO Dane Butcher comments: “Our overriding objective at Symetrix is to make integrators’ lives easier, and it is clear that requirement will gain significance as we move into the world of comprehensive A/V networking. With SymNet Composer 3.0, I believe that we have taken our support of integrators to a new and higher level.”

 

 

LightLead d’Iconix Sound, The World’s First Optical Analogue Musical Instrument Technology

Iconix Sound LightLeadIconic Sound Ltd announced earlier this year the launch of the world’s first optical analog, jack-to-jack musical instrument cable, The LightLead™.

Following its successful debut at the NAMM show where the company exhibited the prototype, they are excited to announce customers can now pre-order the product from their website, with the first 2000 orders receiving a limited edition LightLead™ at a special launch price. Visit : http://www.iconicsound.com

Iconic Sound have now fully patented their revolutionary technology that offers musicians a mind blowing new lead, which has zero capacitance and zero loading resulting in the most precise, perfect, dynamic and crystal clear guitar sound they’ve ever experienced!

Ideal for guitarists, bass players, violinists, in fact any electric stringed instrument where a lead is required. The LightLead™ preserves the exact natural sound and the new product now runs off 1 x AAA battery at each end giving around 9 hours of playing time.
Traditionally cable manufacturers, using basic multi-strand copper cable, have minimised capacitance but have never been able to eliminate it. This has resulted in a loss of high frequencies especially on longer leads.

Iconix Sound LightLeadThe new Patented LightLead™ technology is unique in the existing cable industry using analog fibre optic technology to provide zero capacitance. This achieves a completely flawless analog signal transmission with no compression and faithful reproduction of the instrument tone, whatever the length of lead!

The PVC coated LightLead™ uses ultra high density, small diameter optical analog cables that are resistant to crushing and impact giving them a virtually infinite life.

They are electrically safe with full electro-magnetic immunity, which makes them distortion and interference free. Subsonic and motion noise, static and interference will be a thing of the past; never again will musicians find their cables acting as aerials. With these advantages they are ideal for use in environments that depend on consistent reliable sound delivery whether it’s on stage or in the studio. The future is light!

 

 

New stereo personal monitor system

Shure introduces the PSM300

Shure PSM300Shure announced the addition of the PSM300 Stereo Personal Monitor Systems to their existing line of In-Ear Monitoring (IEM) technology.
PSM®300 introduces the clarity and precision of stereo, 24-bit digital audio processing to the PSM line, while still offering users the customised mix control for which PSM systems are renowned.

The PSM300’s two-tier entry-level/pro-grade control interface allows any user to get a personal monitoring mix up and running with speed and confidence.

PSM 300 will be offered in two system configurations. The baseline PSM 300 Stereo Wireless Personal Monitor System includes SE112 Sound IsolatingTM Earphones and delivers detailed 24-bit digital audio processing and reliable wireless operation over a range in excess of 90 metres.

Shure-patented Audio Reference Companding technology delivers high-quality, low-noise audio, free of artifacts and dropouts. Easy to set up and operate, the PSM 300 will scan, find and assign a clean wireless channel at the touch of a single button, and setting up a personal two-channel mix is easy with the intuitive MixModeR technology.
The PSM 300 Premium Wireless Personal Monitor System offers all of the benefits of the baseline system, but adds high-quality SE215CL Sound IsolatingTM Earphones and a durable aluminium bodypack receiver, optionally with the SB900 lithium-ion rechargeable battery.

The PSM 300 systems and their individual constituent components will be available for purchase from November 1st, 2014.

 

Intelligent power

Lab Gruppen IPD 2400

With the IPD “Intelligent Power Drive” series, Lab.gruppen has created a new concept for the amplification of medium-power sound reinforcement systems that incorporates a DSP platform and intelligent power management into an affordable 1U rack sized package, which is light and easy to set up.
To do this, the Swedish manufacturer has changed its tune in amplification technology by opting for fully bridged class D instead of class TD (tracking class D) and is working on the optimal management of energy resources of the switch-mode power supply (without PFC) by distributing the available energy between the two channels as required.

L’IPD 2400 de Lab gruppen

The IPD series currently consists of two models: IPD 1200 and IPD 2400, which, as their names suggest, are capable of delivering a power of 2 x 600 W or 2 x 1200 W into 4 ohms, with both channels operating simultaneously. As we will see further on, in the tests, this refers to a peak power and not an average continuous power (RMS), but that meets the needs of program material with a crest factor*, depending on the type of music, that varies between 8 dB and 20 dB.

Overview

The front panel

The front is uncluttered, as the features of the front panel are contained in the obligatory, relatively ample area of one rack unit. In addition to the lateral handles for rack mounting, there are two ventilation openings: viewed from the front, the one on the right is the air inlet and on the left is the discharge. At the center is the blue LCD display (not always very visible in daylight, so adjust the contrast), framed on the left by navigation buttons (menu and back) and by the mute buttons (one per channel) with their green indicators.

IPD-frontThese indicators turn red in case of clipping at the input (pre-mixer). To the right of the display are the LEDs that indicate the activation of the limiters and the rotary encoder for selecting and configuring parameters. Pressing this encoder also provides the confirm/set function. The standby button (ON/OFF) with its LED is located under the display.

The rear panel

The IPD series amplifiers have two balanced analog inputs with link outputs and an AES/EBU input (2 channels) also with link (and automatic switch-over). In the case of a complete connection, the AES inputs can take control if something goes wrong or there is signal loss on the analog inputs. This is an interesting operational fail-safe. Similarly, the output connections are via SpeakOn (1+/1-) for each channel but are also available on binding post terminals for stripped wires respecting the Low Voltage Directive. These also accept standard banana plugs.

IPD-rearThe NL4 SpeakOn 1 connector is fully wired for 1+/1- and 2+/2-, with the output of the two channels. All configurations are thus possible, which corresponds to the operating flexibility of these products, suitable for both fixed or temporary installation and touring with small sound reinforcement systems. Be aware that the channel outputs are already bridged with a peak deviation of about 100 V and, therefore, impossible to bridge (so you should not try this).

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On board signal processing

IPDs include a full signal processor with input matrixing (analog and AES), all in 1 rack unit. This should be taken into consideration when comparing the price and the size of a 2-channel amplifier and that of a 4-into-2 processor. The technology used was probably inspired by Lake platforms that are now owned by TC Group, which also owns this Swedish brand. Besides local control via the front panel, Lab.gruppen offers a monitoring software called IntelliDrive that allows the complete configuration of an amplifier and even the formation of groups of amplifiers, which can, for example, be muted, unmuted or placed in standby with a click.

Two-way input mode

Two-way input mode

Stereo mixer input level screen

Stereo mixer input level screen


Each amp is identified by its MAC address and IP address, as the control connection is established via Ethernet, either directly, in the case of a single amp, or via a switch or a WiFi router when multiple amplifiers are networked. Recognition is automatic.

We would have liked a USB port for local configuration because even if the user interface and the depth of the menus are simple, it is always less tedious than manual navigation in the menus and submenus. Of course, we can do the same with a single RJ45 Ethernet cable between the PC and the IPD but there is usually only one Ethernet port on a PC, whereas there are usually multiple USB ports.

IPad app for IntelliDrive

IPad app for IntelliDrive

Also, an embedded Ethernet switch would allow chaining a group of amps without requiring the use of an external switch, which is always limited by the number of ports available. But these are observations that we should make while also considering the price (see below), which is quite reasonable for an amplifier/controller in this class.

By the way, there is indeed an application for iPad (available from the App Store) to control IPD amplifiers, but we have not evaluated it. We recommend that you consult the Lab.gruppen website, where there are videos about its use and about the IntelliDrive software.(http://labgruppen.com/products/ipd_series_touring).

Lab gruppen IPD2400 menu synoptiqueThe above illustration from the Lab.gruppen documentation shows the overview and the functional blocks of the amplifier, the signal path and the menu directory tree when setting it up from the front panel.
When multiple devices are networked, the “global” window allows you to form groups, to name them and the individual devices, to split a group or one or more devices of a group, …
Control from IntelliDrive occurs in real time, settings are applied directly on the amp involved and, conversely, settings made locally through the menus of an amp are instantly reflected in the control software.

IntelliDrive output equalizer (10 bands)

IntelliDrive output equalizer (10 bands)

As for processing, the user has access to input and output gains, a 10-band parametric equalizer at the input and output, shelving, low-pass and high-pass filters, two-way crossover filters with different responses and the possibility to have asymmetric and symmetric filters together and, finally, input and output delays up to 2 s.

The output voltage limiters can be set in volts or in dBu with reference to impedance. These are used in the presets to limit the maximum voltage swing in order to prevent excessive mechanical excursion in the transducers. Other limiters, prior to these in the signal chain, are dedicated to RMS limiting (to avoid overheating moving coils).

Construction

The electronics are spread over several FR4 (epoxy) printed circuit boards, seven in total. These include two dual-mono output boards, which incorporate the power supply and the class D amplification for each channel. These boards are double sided and double layered, populated with THD (through-hole) components on one side and SMD (surface mount) components on the other.

Lab Gruppen IPD2400 Full internal view

Full internal view of the two boards of the amplifier channels on the left, with their integrated power supplies. On the right, you can see the board with the capacitors for energy storage that are shared by the two channels.

Both boards are supplemented by a PCB that we will call “power reserve” board, essentially populated with capacitors, which double the capacity mounted on each of the power-supply/amplifier boards.

The heatsinks consist of two aluminum brackets supporting the amplification boards. Class D operation and the variable-speed forced convection explain this minimal choice, which reduces the amount of material and, therefore, the overall weight. In fact, an IPD 2400 weighs only 6.2 kg.

Even at high power, the internal temperatures never get really high, so the manufacturer was able to choose low-ESR capacitors at 85° C, as opposed to 105° C.

Lab gruppen IPD2400 The input cards

The input cards, conversion, signal processing and network interface.

Lab gruppen IPD2400 The switching H-bridges

The switching H-bridges with their filters, at the bottom left of each aluminum bracket.


Behind the front panel, there is a protected PCB that supports the display and the encoder (and, of course, the buttons). This is connected by ribbon cable to the control (and conversion) board that is located on the rear panel, which also supports the DSP board.  Finally a narrow board provides the output connections: SpeakOn connectors, terminals and mains.

It is all careful workmanship which, considering the modular architecture, suggests that another series or addition(s) could soon be available.
Forced convection is carried out by a pair of low profile fans that draw air from the right of the front side (seen from the front). This air circulates between the heatsink brackets and is discharged on the left of the front and, partially, on the rear.

Measurements

Power and response

With the effective power (RMS) on a sinusoidal signal, using both channels in a stereo configuration, we measure a little more than 150 watts per channel, both at 4 ohms and at 8 ohms. Operating only one channel, the average power rises to just over 320 W into 8 ohms and 295 W into 4 ohms. In fact, it is the power rail limiters that come into play here and provide a constant overall average power.
For this test, we let the voltage limiters peak at their maximum (98 V peak*) using the IntelliDrive software (which can also be done via the front panel), with the objective of measuring the peak power, knowing the limitations involved.

Then we conducted EIAJ burst measurements, with cycles in sequences of 20 ms to Vmax (Pmax) over a period of 200 ms to assess peak power. And, contrary to what is generally practiced, low level outside the peaks was set to average P, 150 W, which we believe is more consistent with the operational reality than at zero.

Figure 1 shows the signal for 8 ohms, built with Audio Precision APx Waveform burst utility, and the response of the amplifier (oscilloscope) in Figure 2, which corresponds to 49 V peak (8 ohms) and 35 V peak (4 ohm) for the lower level and 98 V peak for the high level. The amp goes smoothly through these bursts and delivers 600 W peak (98 V peak) at 8 ohms and 1200 watts peak (98 V peak) at 4 ohms each with a crest factor of 6 and 12 dB, with a THD less than 1% on the peaks, without hard clipping.

IPD2400 burst CF 6dB

Figure 1

IPD2400 TEK02

Figure 2: waveform corresponding to the burst in figure 1 Both channels operating at 8 ohms


Figure 3

Figure 3

Two other ways of seeing the potential behavior are provided in Figures 3 through 5. We highlight the voltage limitation with a voltage ramp in Figure 3.

Mind that, in order for the curves at 4 and 8 ohms not to overlap, we kept the reference load at 8 ohms for power in both cases (so you have to divide it by two at 4 ohms).

We can clearly see that, after peaking, the power stabilizes at 150 W (with both channels operating simultaneously).

Figures 4 and 5 give a temporal representation of two channels (0 dB gain) solicited at a Pmaxmax level by a continuous wave of 500 ms. Note that the P peak lasts longer than 50 ms at 8 ohms as well as at 4 ohms.

Figure 4

Figure 4

Figure 5

Figure 5


Figure 6 is the result at 8 ohms, at -1 dB Pmax. The “high” level lasts almost 100 ms.
The IPD2400 therefore delivers its declared peak power, complying with the most stringent tests that correspond to the realistic use of percussion, where the attack and resonance will be reproduced well even at low frequencies.

Figure 6

Figure 6

The intelligence of the power management consists of allocating all or part of 300 W RMS for each channel as needed while retaining the possibility to reproduce peaks at Pmax.

Although they can work in stereo, IPD amplifiers would more advantageously – as it is their primary purpose – be used in two-way systems (or in three-way systems with passive mid-high crossovers), with the crossover filters, EQ corrections and the delays they provide.

To conclude this section about power, note that the IPD 2400 supports 2 ohms loads, although at lower RMS power (2 x 70 W at 0.5% THD), due to the current limiters, and deals perfectly with short-circuits (see Figure 7).

Figure 7

Figure 7: short circuit with a square wave at full power

Finally, we performed measurements on a complex load by placing a bank of high voltage capacitors totaling 132 µF in parallel with our measurement load, corresponding to 8 ohms at a load impedance of 4 ohms modulated at 250 Hz and with a current/voltage phase shift of 60° (Π/3).

Full power at 250 Hz is achieved without degradation and with the same THD as into a purely resistive load.

This is not surprising with a bridged class D structure, where the amplification (switching) stages recycle their reactive power into the power supply without pumping.

The load used and the audio filter, above on the left.

The load used and the audio filter, above on the left.

It’s one of the benefits of this amplification structure that make it ideally suited for the highly reactive transducers with high BL specs that are now available.

The frequency response is shown in Figure 8, where we see that the high cut-off is mainly due to the conversion and processing at 96 kHz, which infers a steep cut just below 48 kHz.

In Figure 9, the response is due to our interposed measurement filter that cuts around 30 kHz with a 6th order Bessel response.

This filter is essential for distortion and dynamics measurements but does not alter the phase in the audio band. We see that the slight overshoot, due to class D output filters and negative feedback, has disappeared.

Figure 8

Figure 8

IPD2400 reponse frequence filtre

Figure 9


Input level, gain, CMRR and damping factor

The maximum analog input level is +20 dBu with a balanced input impedance of 18 kΩ. The gain of the amplifier (with the input and output gain of the processor at 0 dB) is about 36 dB and the peak power is obtained at +6 dBu input in analog mode and at -14 dBFs in AES mode. This means that working at +4 dBu (usually 0 dB) on the console or preamp, and allowing for peaks of 12 dB, the overall gain should be set at -8 dB, at the most.

To obtain the best dynamics, you should, instead, lower the output gain (the one directly accessible from the encoder outside the menus). We measured an output impedance of 17 mΩ – very nice – which, with an 8 Ω load, provides a damping factor of over 470. Bass transducers are well controlled, with a minimal lag. This measurement takes into account the contact resistance of the connections and thus represents real-world field conditions, where you also have to take into account the length of the connecting cable and its resistance per unit length.

The rate of common mode rejection on the analog inputs is constant at the three frequencies (40 Hz, 1 kHz and 10 kHz) at 68 dB on one channel and 58 dB on the other. This is a fairly large disparity between the channels, but the manufacturer declares greater than 50 dB. It is therefore entirely consistent. This measure is less important on a power amp than it is on console inputs, as the lengths of connection cables and the sources are not the same.

Distortion and dynamics

For this kind of measurement on a class D amp, a filter is inserted to attenuate fast-rising HF signals due to the switching process (see above) that can disturb the input circuits of the analyzer. Obviously, this limits the THD measurement with all harmonics up to the 10th order around 3 kHz. At 10 kHz, the only ones to be taken into account are the 2nd and 3rd harmonics, which are, despite everything, still predominant.

Figure 10

Figure 10

In Figure 10, where the curve is taken at 2 x 100 W into 4 ohms, and where we have intentionally expanded the frequency sweep, we see the abrupt break at about 16 kHz where the 3rd harmonic falls.

The results are very good for a bridge-operating class D amp and, above all, the two channels follow well, indicating good manufacturing uniformity.

In fixed frequency mode, we found harmonic distortion rates of 0.002% at 40 Hz, 0.008% at 1 kHz and 0.012% at 5 kHz at 130 W. This is quite decent. Intermodulation distortion (SMPTE, 60 Hz, 7 kHz at a 1:4 ratio) at 2 x 150W is around 0.01%. And when the RMS limiter threshold (knee) is reached, the harmonic distortion does not exceed 0.1%, which is remarkable.

As dynamics are concerned, we measured a signal to noise ratio of 95 dB (un-weighted) and 97.5 dB (A) at -22 dBFS using the AES input, which corresponds to a S/N ratio at maximum level of 104 and 106.5 dB respectively, which is excellent for a class D amplifier in particular and very good in general.

Figures 11 through 13 show an FFT (Fast Fourier Transform) frequency breakdown, with a stimulus at 1 kHz and a 2 x 100 W output (at 8 ohms).

Figure 11

Figure 11

In Figure 11, the measurement filter is not inserted to highlight the lines of the PWM. We see that the frequency, fixed here, of the pulse width modulator is about 350 kHz (far from the audio band), but still at a level of -6 dBV (0.5 V).

We understand that this could create measurement artifacts. Fortunately, this is inaudible and totally compensated by the loads representing the speakers.

In Figure 12, the same FFT is taken with the measurement filter inserted. The principal PWM peak is below -60 dBV and the HF noise has almost completely disappeared.

The good news is that the audio spectrum is almost identical, indicating that there are no further intermodulation products, as can be seen in Figure 13 where we have limited high frequency range in order to enlarge the details on the x-axis.

IPD2400 FFT avec filtre

Figure 12


IPD2400 FFT 1K Figure 13[/caption]


Nevertheless, there is a peak at 650 Hz with harmonics 3 and 5, which is probably due to a beat with a multiple of the mains frequency, at a level almost identical to that of the second harmonic of the stimulus.
We also note that the 5th harmonic is predominant at 8 ohms and the 3rd at 4 ohms (Figure 13), which is due to the idle time that is inevitably introduced in the control of the half-bridges (MOSFETs) of the class D output to prevent any simultaneous conduction of this high switching frequency, and to the LC output filter that is better suited to a 4 ohm load.

Figure 14

Figure 14

Figure 14 shows the response to a square wave at 1 kHz with one channel operating at 312 W RMS (8 ohms). The front edges are clean and the over-oscillations are due to the digital filtering with a pseudo-frequency located at less than Fs/2 (48 kHz).

The excessive oscillation stays within the order of 10% of the peak-to-peak voltage. Rise times and fall times are similar with a traverse rate of about 40 V/μs for the amplification stages.

Latency, with conversions and signal processing performed at 96 kHz (24 bit quantization), is in the order of 620 or 660 microseconds depending on whether we drive the amplifier with an analog or an AES input.

Figure 15

Figure 15

One might expect less with a digital input but, in fact, an SRC (Sample Rate Converter) is always inserted (Figure 15).

In Figure 16, we added a delay of 11.628 ms on channel 2 via the IntelliDrive control interface, which is thus addition to the inherent latency of 620 μs. The time entered corresponds perfectly, as we obtained a total of just over 12.2 ms.

Finally, in Figure 17, the IPD 2400 is operated in 2-way mode with a 4th order Linkwitz-Riley type crossover filter set at 800 Hz. We also applied corrections to the low and high bands in accordance with the associated screen shots. As seen on the curves in Figure 17, everything corresponds perfectly. IPD allows you to save one hundred presets per library. This is more than you would ever need, even if you had a huge number of loudspeakers or of speaker systems of all types.

Figure 16

Figure 16

Figure 17

Figure 17


Input EQ (Figure 17)

Input EQ (Figure 17)

4th order LR crossover at 800 Hz (Figure 17)

4th order LR crossover at 800 Hz (Figure 17)


The power supply

The universal power supply (100-265 V) or, rather, the two power supplies (one per channel) do not feature power factor correction, as demonstrated in Figures 18 and 19, taken at half power and Max RMS power (300 W) respectively, which show that the peak current exceeds 10 A (voltage and current probes 1/100).

Figure 18

Figure 18

Figure 19

Figure 19


Note the crest factor of the current is in the order of 5, which leads to an RMS current of approximately 2 A. This is typical of direct rectification, in which the current wave corresponds to the conduction of the rectifier diodes at the peak of the sinusoid wave. The current corresponds to the blue trace, while the voltage is shown in yellow.

Peak current in stand-by

Peak current in stand-by

This screen shot form the oscilloscope shows the current draw in stand-by; the peak current drops to 500 mA, which, together with the high crest factor, results in a power consumption of about 10 W. Only the auxiliary supplies are functioning to power the control electronics, including the network interface.

Conclusion

The IPD series and, in this case, the IPD2400 are complete amplifiers that will adapt to all types of medium-power speaker enclosures and to installed sound systems.
The on-board signal processing is comprehensive and easily configurable, and the performance is up for the job. In addition to being controlled through a wired Ethernet network, they can now even be controlled remotely with a wireless router and a tablet using the Lab.gruppen application.

We particularly like the intelligence of their power management, their compactness and low weight. In short, they feature smart power and a high quality/price ratio for a European design and quality manufacturing in Thailand (with the European standards).

* Crest Factor : ratio (which can be expressed in dB) of the peak value of a parameter (power here) and its average value, therefore 10 log(Ppeak/Prms). Musical signals, depending on the type of music, have a crest factor ranging from 6 dB (highly compressed music) to more than 20 dB (symphonic music), approximately 12 dB on average.

* * 98 V peak corresponds to a peak power of 982/16 at 8 ohms 982/8 at 4 ohms, with a sinusoidal signal – respectively 600 W and 1200 W.

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Meyer Sound Launches User Training Videos for LYON Line Array System

Tutoriels Meyer SoundMeyer Sound has released four online video tutorials to support users of the LYON™ linear sound reinforcement system, the newest addition to its LEO™ Family of line array products.

Covering topics that range from system design and rigging to optimization.

The video modules are available for viewing on the company’s video page :

http://meyersound.com/news/2014/lyon_video_tutorials/

MEYER PLS14 LYONLYON Product Introduction : The comprehensive overview outlines the system’s coverage patterns, rigging features, connections, and integration with the 1100-LFC low-frequency control element and the Galileo® Callisto™ loudspeaker management system.

LYON Array Assembly : The video gives a step-by-step tutorial for presetting splay angles, lifting and locking an array, and pullback techniques. It also shows the rigging calculator features in the MAPP Online XT™ acoustic prediction software.

LYON Compass Presets and Controls : The module shows the user workflow for optimizing a LYON loudspeaker array, featuring presets in the virtual Galileo Callisto array processor, array configuration in zones, integration of 1100-LFC loudspeakers, and the U-Shaping™ filters for output equalization.

Meyer PLS14 Lyon riggingUsing Array Correction and Delay Integration : The video demonstrates the use of array processing and correction and delay integration for Meyer Sound line array systems.

These training resources complement Meyer Sound’s industry-leading education program, which includes in-person seminars and online resources for all aspects of sound reinforcement, from audio theory and mixing workshops to in-depth courses on the design and optimization of complex systems.

Learn more about the Meyer Sound education program here: http://www.meyersound.com/events/seminars/calendar.php

A calculator app for tablets and smartphones

PAcalculate from Brusi Acoustics

brusi consultingBrusi Acoustics, a consulting firm based in Valencia (Spain), has announced the availability of PAcalculate, a free multi-lingual multi-platform mobile app with calculators and other information and utilities for pro-audio and lighting professionals. PAcalculate is lightweight and can be installed on iOS (Apple), Android, Windows-Phone 8 and Blackberry devices from their respective app stores.

The application uses responsive design developed to work on screens that range from a classic Blackberry to a large tablet. Both metric and English units are supported.

PA related calculators include SPL, SPL addition, ceiling speaker coverage, air absorption, time to distance (and vice versa), frequency to wavelength (and vice versa) with full and half lambda options for arrays, Q factor to bandwidth conversion, dBu-dBV-Volts and speaker cable losses (low and high impedance). For lighting, a DMX calculator as well and RGBW and CMY LED emulations are provided.

Some examples of screens with different functions and languages :

PAcalculate EN SPLPAcalculate Frecuencia

PAcalculate Speaker cable EN


PAcalculate menu-iPhone4-SpanishPAcalculate-EN-RGB-RGBW-RGBAPAcalculate-EN-Recording-file


Calculators provide comprehensive options as well as built-in help. Other calculators are not directly related to the industry, but will nonetheless be useful: Ohm’s / Joule’s Law, KVA, BPM to time and conversion of weighs and measures.

PAcalculate-EN-DMX-BlackberryIn addition to calculators, pin assignment information is provided for some of the most common audio and lighting connectors. Lastly, an inclinometer utility rounds up the feature set.

At that time, PAcalculate came with English, Spanish, Portuguese and Simplified Chinese language sets. The developers are seeking collaborative translations to other major languages, translators will be listed on the credits. The company estimates that the app, which can be used without an Internet connection, will be installable in 94% of smart phones.

“We tried to pack an extensive bunch of tools that could be useful for sound reinforcement professionals doing live or fixed install work, plus some lighting ones too. While there is no rocket science involved in the calculations, some can get cumbersome, specially in the field, and others can get very complicated very easily (such as calculating time delays taking into account the effect of temperature, which can be critical for delay synchronization).

Others, like KVA calculations or unit conversions are not specific to our industry, but will nevertheless be useful to have under the same roof, without the need to exit the application. We’ll be listening carefully in the future and adding new tools and contents based on users’ feedback”, comments Senior Consultant José Brusi.

For further information : http://www.brusi.com

Sound Tutorial Videos for CAL Column Array Loudspeaker and MAPP Online Pro

To expand its training support for product users, Meyer Sound has released two new online tutorial videos, focusing on the use of the MAPP Online Pro® acoustical prediction program and the Compass® control software for the CAL™ column array loudspeakers.

These videos are available on the Meyer Sound website and on YouTube

The CAL tutorial shows how to use the Compass control software to set up and operate the CAL loudspeakers and control their beam steering and splitting capabilities. Viewers are given an overview of initial system configuration, input selection, preset selection, and parameter modification for creating new presets.

The MAPP Online Pro tutorial introduces users to the software’s main operating modes, and guides them through the basic features of the program, and its use in loudspeaker coverage prediction.

For the Québec summer festival

HI-Roof : An outsized aluminum mobile stage

On request of the Quebec City Summer Festival, Unison Structures, a Quebec company specialized in the manufacturing of aluminum trusses and architectural elements, achieved the largest freestanding mobile stage designed so far in North America, called Hi -Roof.

View of the Bell (Hi-Roof) stage at the Quebec City Summer Festival during the Def Leppard show (picture by R. Philippe).

View of the Bell (Hi-Roof) stage at the Quebec City Summer Festival during the Def Leppard show (picture by R. Philippe).

The first official setup of the stage took place on June 7 (to 17) on the Plains of Abraham park (historical site), where the festival runs. Since then, the stage hosted performers Bruno Mars, Stevie Wonder, The Black Keys, Def Leppard, Rush and many others during this 11-day festival (between July 4th and 14th) and more recently those of Paul McCartney, July 23, and Celine Dion, on the 28th.

According to Patrick Martin, Production Manager of the Festival: “We really needed to expand our main stage (Bell stage), to improve its technical capabilities while increasing safety. Unisson has presented an innovative concept meeting all these criterias”.

For Olivier Jobin, Vice President of Unisson structures: “We are proud to offer so far, the stage with the highest degree of technological achievement. Working with the leaders of the Quebec City Summer Festival expertise, we have been at the heart of the issues of our more sophisticated customers. “

Hi-Roof

The Hi-roof Stage (patented), which took five years of R&D and hard work, stands out for its size and capabilities with unusual and impressive values :

  • The total area footprint is about 68 m (width) by 32 m (depth) for a height of 26 m.
  • The stage itself has an scenic area of ​​about 500 m2 with an aperture of more than 27 m.
  • Wind resistance is 120 km/h, and the stage can handle loads of 730 kg per m2.
  • The total hanging capacity is 68 tons for a uniformly distributed load!
Map of Hi-Roof stage with dimensions. The structure supports gusty winds of 120 km / h and carries 68 tons hanging!

Map of Hi-Roof stage with dimensions. The structure supports gusty winds of 120 km / h and carries 68 tons hanging!

Ideal for orchestral recordings

Sennheiser MKH 8090

With the new MKH 8090, wide cardioid directivity microphone, Sennheiser widens its studio recording microphones range. Located between the omnidirectional MKH 8020 and the cardioid 8040, the MKH8090 is “the best solution for orchestral recordings” according to Kai Lange, Sennheiser wired microphones product manager. “Used as main microphone, it captures the globality of the sonic scene with the right amount of room acoustics. When used in near field, it offers enough directivity to get rid of other sonic sources without offering a too mate image.

The MKH 8090 shares all accessories of the 8000 range, like different height stands, holders, suspensions, different length cables, windshields… and as its predecessor the MZD 8000 digital module instead of the XLR module. This module converts the MKH 8090 audio signal into an AES42 (mode 2) digital signal right after the microphone capsule. The MKH serie is based on the HF condenser principle which was developed by Sennheiser over 50 years ago and fully mature today with the use of symmetrical transducer technology.

Four-part element cardiod condenser microphone

Audio-Technica AT5040

l'AT5040 avec sa suspension dédiée AT8040During the last AES convention in San Francisco (October 26th to 29th) AudioTechnica introduced the serie 50, a new range of very high quality studio microphones and presented its first model the AT5040, a vocal cardioid condenser microphone with exceptional characteristics: 5 dB of noise, 137 dB of dynamic range (142 dB maximum level with 1% THD), -25 dBV at 1 Pa (56 mV) sensitivity and a frequency response of 20 Hz to 20kHz within less than 2 dB.

To reach this goal, Audio-Technica uses a patented system : four matched extremely thin membranes (2 microns) with summed outputs bringing the advantages of a large membrane (sensitivity and dynamics) without drawbacks : increased mass creating a loss of high frequency response and transient degradation. The equivalent surface of this combination, reaches the double of a one inch capsule.

Vue interne de l'AT5040 Audio-TechnicaThe AT5040 is completely hand made with selected discrete components. The capsule is completely decoupled from the aluminum-brass chassis with an internal suspension. Audio-Technica has designed a completely new and patented suspension, the AT8040, which eliminates any resonance and filter completely all mechanical noise.

If this microphone was primarily designed for vocal recordings, it will be fully recommended when recording acoustical instruments such as piano, guitar, strings or even saxophone with its great sonic purity and its excellent transient response.

The AT5040 will be available in January 2013 and its retail price will be of 2990 €.

Specifications :

Capsule : permanently polarized condenser, cardiod

Frequency response : 20 – 20 000 Hz

Impedance : 50 ohms

Sensitivity : – 25 dBV (56.2 mV) à 1 Pa (94 dB SPL)

Maximum SPL : 142 dB (1% THD, 1 kHz)

Noise : 5 dB SPL

Dynamic range : 137 dB, 1 kHz (SPL max)

Signal to noise ratio : 89 dB (1 Pa)

Phantom power supply : 48 V DC, 3,8 mA

Mass : 582 g

Connector : 3 pin XLRM