Products of Interest
AudioKit Audio Developer
Framework for iOS Apps
AudioKit is a free, open source frame-work for adding audio synthesis, pro-cessing, and analysis features to iOS, Mac OS, and Apple Watch apps. It in-cludes synthesizers, physical models, sample and playback functionality, effects, filters, delays, reverbs, and other processing options. A range of sample applications are provided, in-cluding pre-made instruments, audio players, samplers, synthesizers, and CoreAudio sequencers.
The code is written in Swift but the audio processing is carried out in C and C++. The user can quickly develop audio applications in Xcode Swift Playgrounds and 150 example playgrounds serve as interactive tuto-rials. Unit and continuous integration tests have been incorporated into the framework.
AudioKit can be integrated with CoreAudio, AVAudioEngine, Syn-thesis Toolkit (STK), and Sound-pipe/Sporth audio engines, the latter of which can include code from Chuck, CSound, Faust, Guitarix, and ToneStack. The toolkit is available for free under the MIT License.
The AudioKit Web site includes a range of audio apps that have been created with the framework. The most recent of these is Synth One, a free and open source synthesizer for iOS. It is the result of a collab-oration of over 100 volunteers from developers, to sound designers, and musicians. It has five oscillators (two digitally controlled, FM, sub, and noise), and two assignable low-frequency oscillators. More than 100 alternative scales and tuning options are provided. The synthesizer also has a vintage-style 16-step sequencer, a polyphonic arpeggiator, a four-pole vintage low-pass filter, a high-pass doi:10.1162/COMJ r 00491
filter, multi-tap delay, and touchpads. There is a MIDI input for use with external MIDI hardware or a DAW, and MIDI learn is available on all knobs. Audiobus 3 and Inter-all Au-dio (IAA) are supported. The source code for the synthesizer is available to users.
Contact: Audio Kit; Web audiokitpro.com.
Sound Design Toolkit
The Sound Design Toolkit (SDT) is a collection of sound synthesis models that are based on physical descriptions of how real-life sounds are produced. They are designed to generate both basic and more complex sounds based on the inter-action between solid objects, liquids, gases, and machines. The models are implemented by procedural audio algorithms that use computational descriptions of the event that pro-duces the sound, rather than sample-or wavetable-based techniques. The models are intended to be intuitive to control.
The core library of the SDT is written in ANSI C, on top of which there are a set of wrappers for Max and Pure Data (Pd), which access the SDT framework through externals. A set of Max patches makes up the final top layer and includes a GUI and help files. The C API that makes up the core of the system can also be used independently. The toolkit is designed in a modular way and the basic modules can be combined to create complex interactions—for example, various friction models with resonators.
The toolkit can produce four main types of sound: vibrating solids, liquids, gasses, and machines. Al-gorithms for the interactions with solids use a structure that represents an interaction between two resonat-ing objects. An inertial mass is a
simulation of a basic inertial point mass and most commonly serves to excite a modal resonator, which is a physically modeled set of mass-spring-damper oscillators. Impact and friction interactors are provided. The impact interactor is nonlinear and cal-culates the force based on the relative displacement between the contact points and the friction interactor uses the relative velocity between them. More complex interactions with solids, such as rolling, crum-pling, breaking, and scraping sounds, can be created using stochastic series of impacts with different time and amplitude values. Three compound algorithms for interacting with solids are provided in the toolkit and their output can be used to control the single impact model.
For liquids, the sound of gas trapped in bubbles is simulated using a polyphonic bank of exponentially decaying sinusoid oscillators with a stochastic control algorithm. The main model used with gasses is turbu-lence, with the obstacle that causes the turbulence simulated. Three types of obstacles are included: noise (solid surfaces), howling (cavities), and Karman vortices (thin objects). Explosions are modeled using a Fried-lander waveform, which delivers a sudden high pressure with a negative expansion tail.
The machine sounds are divided into combustion engines and elec-tric motors. The combustion engine is modeled by four periodic wave-forms that represent the intake and outtake valves, piston, and fuel ig-nition. These are used as input or feedback gain modifiers for digital waveguides that model the exhaust system. The electric motor is mod-eled as the sum of sinusoidal partials for the pitched tones of a rotor. The sound of motor brushes are created by using these partials as amplitude en-velopes for filtered noise. Mechanical stress and imbalances are simulated
by frequency modulation. Chassis resonances are simulated by a comb filter. Finally, noise from rotating parts and the fan is simulated by continuously filtered noise.
A further set of sound-processing algorithms, such as pitch shifting and reverb, are included, and audio feature extractors, such as a frequency esti-mator and zero crossing rate, are also provided. The GUI controls allow the user to easily adjust the parameters of the model they are using and each comes with a help patch. Example patches that demonstrate compound models are also provided with the toolkit. A number of videos showing sample applications are provided on the toolkit Web site.
The SDT source code is freely available and is licensed under the GNU Public License. Contact: Web www.soundobject.org/SDT/.
Tonstrum Traveler Doppler
Effect Plug-In
The Traveler plug-in from German company Tonsturm is a plug-in for Macintosh computers that models the Doppler effect using 64-bit processing. The user can control the duration of the Doppler path, the speed of the object/sound source moving along it, and the length of the path. The duration can be specified in seconds, bars, beats, or timecode. A graphical path editor uses a red dot to indicate the listeners’ position and it can be freely placed relative to the Doppler path. The user can double-click on the grid interface to add new nodes to the path, which can be dragged around to alter its shape. A yellow dot is used to show the position of the sound source on the path. A manual mode is also available and it allows the user to move the sound source along the Doppler path in real-time using a large dial.
A loop and granular sample player is also included in the plug-in. The granular sampler can be synchronized with the Doppler path. The user can adjust the grain, pitch, size, density, pan, and position, and can add a jitter randomization element to these values.
There is also an effects section, with tremolo, delay, distortion, equal-izer, and compression. Four multiseg-ment low-frequency oscillators can be used to modulate the granular sample player and the effects parameters. A MIDI learn feature is also included and can be used with almost any parameter in the plug-in to allow it to be controlled from an external device. The plug-in is designed to operate as a live insert effect. It includes more than 80 presets. It is currently only available for Macintosh computers in VST, AU, and AAX formats, but a Windows version is in development.
The Traveler plug-in is listed for US$ 229. Contact: Tonsturm; e-mail [email protected]; Web www.tonsturm.com.
Acon Digital Acoustica 7
Acoustica 7 is a software application for editing, mastering, and restoring audio on Macintosh and Windows machines. The application supports sampling rates up to 384 kHz and sample depths of 32 bits. It can be used to edit 5.1 and 7.1 surround sound.
Processing tools from Acon Digi-tal’s Equalize and Verberate plug-ins are included in the application, along with reverb, zero latency convolu-tion reverb, echo, modulate, chorus, time-stretching, pitch-shifting, and a harmonizer. Chains of audio pro-cessors can be created and saved for future use. A brick-wall limiter with a pre-compression stage is available and there is a combined multiband
compressor/expander/gate with sep-arate processing possible across four bands.
The user can control the crossover frequencies and filter slopes (from 6 to 96 dB per octave) of the dynamics processor. Both the limiter and dy-namics processor have an adjustable look-ahead and an optional up to four times oversampling feature. A new dither tool, with flexible noise-shaping features, has been added.
Acoustica 7 also includes the audio restoration tools from the company’s Restoration Suite, including DeClick, DeClip, DeNoise, DeHum, Vitalize, and automatic removal of DC offsets. A new spectral editing mode allows the user to switch from a waveform to spectrogram view to select specific time or frequency regions for pro-cessing. Brush, freehand, and magic wand selecting tools are available. A retouch tool can replace noise in a signal based on a selected section of nearby noiseless reference audio.
Audio from different tracks can be mixed in real time and cross-fades created in multitrack sessions. Clips can be looped or time-stretched, and volume, pan, and send levels automated. A batch processor is also provided.
The analysis tools in the applica-tion include a spectrum, spectrogram, wavelet analysis, spectral histogram, statistics, and automatic track split-ting. EBU R128- and ITU-R BS.1770– compliant loudness metering and normalization have also been added to this latest version.
Acoustica 7 supports plug-ins in VST, VST 3, and AU plug-ins formats. DirectX plug-ins, which were sup-ported in previous versions, are no longer compatible. The software can import audio files in WAV, WAV64, AIFF, MP4, MP3, OGG, FLAC, and WMA formats. CD burning tools are included. The user interface of Acous-tica has been completely redesigned
Figure 1. The VocalSynth2 from iZotope.
for this release, with vector-based graphics for high-resolution displays.
Acoustica 7 Premium Edition is listed for approximately US$ 249 and the Standard Edition for US$ 74. Con-tact: Acon Digital, Sørkedalsveien 273, NO-0754 Oslo, Norway; e-mail [email protected]; Web www.acondigital.com.
iZotope VocalSynth 2
iZotope has announced a new release of their vocal processing and effect plug-in, VocalSynth. VocalSynth 2 gives the user five vocal tools they can blend between: Biovox, Vocoder, Compuvox, Talkbox, and Polyvox. It also includes hundreds of presets for
immediate use and a newly designed GUI with more detailed controls than previously available (see Figure 1).
The new Biovox module adds vocal characteristics to any audio signal by modeling the human vocal tract. It can add vowel-like sounds and breath-related sounds such as nasal tones or whispers to a track. It also includes pan, filter, and other signal-shaping controls. The Vocoder has Smooth, Vintage, and Hard modes, and ten presets are provided. The user can fine tune the controls using a Scale and Shift knob. The Compuvox tool gives a computerized sound to a vocal track, and can be used to create subtle inharmonic effects or experimental sounds. It has Spell, Read, and Math modes of operation,
which each create different tones. Bits, Bytes, and Bats controls can be used to add digital aliasing noise, prolong vowel sounds, and to give a guttural sound to the vocal. Talkbox is an emulation of vintage Talkboxes. The user can choose between Dark, Classic, and Bright modes, with different frequency responses to create darker, highly articulate, or balanced sound. A cabinet emulation is also included to control the physical characteristics of the speaker and the user can adjust the Drive/input gain and the amount of speaker character. Finally, Polyvox is a polyphonic pitch shifter that can be used to create a natural-sounding choir or group of backing singers, or to add multiple pitch-shifted instances to a sound to
create unnatural and experimental sounds.
VocalSynth 2 also includes seven different stompbox-style effects in a drag-and-drop chain: distort, filter, transform, shred, delay, ring mod-ulation, and chorus. The plug-in is compatible with Windows 7–10, and with Mac OSX 10.8.5–10.13. It is available in AU, AAX, RTAS, DPM, VST 2, and VST 3 formats. The visu-alizations used in the display require a graphics card with Open GL 2.0 support.
Vocal Synth 2 is listed for US$ 199 and it is also available as part of iZotope’s Creative Suite for US$ 499. Contact: iZotope; Web www.izotope.com.
MAGIX Sound Forge 12
MAGIX has released an update to Sound Forge, their mastering, sound design, and editing application. It now features 64-bit architecture and can facilitate thirty-two channel recording at sample rates up to 384 kHz, with 64-bit sample depth. Multichannel audio can be opened, edited, converted, and exported for use in other applications.
Sound Forge now supports VST3 and SEQUOIA effects. POW-dithering algorithms, a set of commercial dithering algorithms, are also in-cluded in this update. Support for DSD file format has been added and the user can choose to import from, and export to, DSD.
A set of new audio effects plug-ins have been added. Wave Hammer 2.0 is a mastering tool with a classic compressor and a volume maximize. DeClipper, DeHisser, and DeClicker are provided for removing distorted clips, low-level white noise caused by audio equipment, and individual clicks and pops. The update also adds RX Elements and Ozone Elements,
two mastering tools from iZotope. RX Elements is a set of basic re-pair tools that includes real-time noise-reduction plug-ins and an audio editor with spectral editing functions. Ozone Elements uses a combina-tion of professional presets, macro controls, and a Mastering Wizard, to assist the user in creating their final master.
Scripting is now supported in Sound Forge for automation of mon tasks and the software is com-patible with SpectraLayers Pro 5.
The Sound Forge 12 Suite includes the new version of Sound Forge, along with MAGIX’s essentialFX Suite, their Analogue Modelling Suite A, Vintage Effects Suite A, VariVerb II, and Vandal guitar and bass amplifier plug-in collections.
Sound Forge Pro 12 is listed for US$ 399 and the Pro Suite version for US$ 599. Three- and twelve-month versions of the ap-plication are also available from US$ 14.99 per month. Contact: MAGIX Software GmbH, Quedlin-burger Straße 1, 10589 Berlin, Ger-many, e-mail [email protected]; Web www.magix.com/us or MAGIX Computer Products International Corporation, 280 Greg St., Suite 10, Reno, Nevada 89502-2272, USA; e-mail [email protected]; Web: www.magix.com/us.
Lekholm Instruments Digital
Harmonica Controller
The DM48 is a 12-hole, 48-note chro-matic MIDI harmonica controller. The harmonica has twelve pressure sensors that work independently and that can detect both blows and draws. The controller can be set to monophonic mode, which allows for only one note to be triggered at a time, or to polyphonic mode, where
multiple notes can be simultaneously triggered.
The instrument sends a MIDI note on message when a note is triggered and then uses MIDI control change messages to modulate the intensity of the sound for the entire duration after the initial note trigger. Pitch bends are also created in a similar way with the DM48. The air pressure is measured at each sensor and bending of notes can be achieved by blowing or drawing strongly. The threshold and sensitivity of the blow for bends can be adjusted by the user.
The harmonica has adjustable response parameters, breathing re-sistance, and customizable tunings. It is powered through a USB cable, which also serves to connect it to hardware and software synthesizers, and is used for firmware updates. The controller has an OLED display for feedback and adjustment of settings. The mouthpiece is constructed from anodized aluminum.
Lekholm Instruments recommend using physical modeling synthesizers with the harmonica, such as Yamaha’s VL70-m monophonic synthesizer and the Acoustic Electro Waves synthesizer for Reason.
The DM48 is listed for US$ 720. Contact: Lekholm Instruments; e-mail info@lekholminstruments .com; Web www.lekholminstruments .com.
IK Multimedia UNO Synth
UNO Synth is the first hardware synthesizer from IK Multimedia. It is designed to be an affordable portable, monophonic, analog synthesizer (see Figure 2).
The all-analog path has two in-dependent voltage-controlled oscil-lators, a noise generator, a resonant multimode voltage-controlled filter, and voltage-controlled amplifier. The
Figure 2. IK Multimedia’s UNO Synth.
voltage-controlled oscillators have options of saw, triangle, and pulse waveforms with continuously vari-able shape, including pulse width modulation of the square wave. A two-pole analog resonant sweepable multimode filter based on Operations Transconductance Amplifier tech-nology provides low-pass, high-pass, and band-pass options. A dual-stage overdrive provides input for the filter overdrive. A low-frequency oscilla-tor offers a choice of sine, triangle, square, up saw, down saw, random, and sample-and-hold options, which can be used to modulate pitch, filter, amplitude, and continuous oscil-lator wave shapes, including pulse width modulation. A built-in delay effect can be synchronized with the synthesizer’s tempo settings.
The top panel of the synthesizer has more than 40 controls for
real-time adjustment of parameters. The 27-note keyboard provided on the synthesizer includes multitouch functionality. It can be used as a chromatic keyboard, set to one of 13 predefined scales, or to a newly defined scale. There are ten different styles available on the four-octave range arpeggiator. Five modulations for performance are also available: Dive, Scoop, Vibrato, Wah, and Tremolo. A 16-step sequencer with 20 automatable parameters are also included.
There are 100 presets included with the synthesizer. Each of them includes settings for the sound pro-gramming, along with an associated arpeggio and sequence. Eighty of these presets can be reprogrammed by the user. A software editor allows the user to store and manage an unlimited number of additional presets.
The UNO Synth can be powered by four AA batteries or by USB power. It has an audio input for use with external audio devices, MIDI input/output, and a USB MIDI port for use with computer. The synthesizer weighs 400 g and measures 25.6 × 15 × 4.9 cm. Purchase includes a micro USB to USB cable, two MIDI jack to DIN cables, and batteries.
The UNO Synth is listed for US$ 199. Contact: IK Multimedia, Via dell’Industria 46, 41122 Modena, Italy, or 590 Sawgrass Corporate Pkwy., Sunrise, Florida 33325, USA; Web www.ikmultimedia.com.
Modal Electronics Skulpt
Polyphonic Synthesizer
Skulpt is a portable, low-cost, portable polyphonic synthesizer from UK
Figure 3. The Skulpt polyphonic synthesizer from Modal Electronics.
company, Modal Electronics (see Figure 3). The synthesizer is based on the architecture used in Modal’s CRAFTsynth and it offers the user four polyphonic synthesis voices. It measures 10× 5.5 × 3 in and weighs less than 2 lbs. It can be powered by USB or by six AA batteries, which can give up to 12 hours playing time. The synthesizer has a 16-key keypad on the front for note input. Located behind this on the top panel are controls and 13 encoders (with a shift mode) for access to various functions. These transmit control change messages for use with external devices and software applications.
Each voice of the synthesizer has eight oscillators available with smooth morphing waveforms, giv-ing 32 in all. There are frequency, ring modulation, and pulse-width modulation options, with eight mod-ulation slots featuring eight sources and 37 destinations. Envelopes for filter, amplitude, and modulation are
provided. Two low-frequency oscil-lators (a global oscillator and one per voice), can be synchronized with MIDI. The modeled resonant voltage-controlled filter has two poles and can be switched between low-pass, band-bass, and high-pass filter.
The synthesizer features a se-quencer with real-time polyphonic note input, which is capable of stor-ing 128 notes, four rows of parameter automation, and 64 sequences. An Arpeggiator with parameter control of the division, direction, swing, and octave range is also included.
Connections on the rear panel include 5-pin for MIDI input, output, and thru, a class-compliant USB port for MIDI output to computer and mobile devices, a mini-jack port for analog sync input and output, a mini-jack master output, and a dedicated headphone output. The synthesizer is compatible with Mac OSX, Windows, iOS, and Android operating systems. A software editor gives the user
visual feedback of and access to edit settings.
Skulpt is listed for US$ 300. Contact: Modal Electronics; Web www.modalelectronics.com.
Luke Audio AL-Y56
Luke Audio’s AL-Y56 microphone is a large-diaphragm, tube condenser microphone that comes with three in-terchangeable capsules, each of which offers different sonic characteristics. The microphone is designed for pro-fessional and home studio use, along with broadcast and live applications. The AL-Y56 features an Electro-Harmonix 6072a dual triode, vacuum tube, an 11.5:1 output transformer, and a 10-dB pad. It has three swap-pable capsules with large 1-micron thin platinum-sputtered Mylar di-aphragms. They combine with the microphone body to create a pres-sure gradient condenser microphone
with a cardioid polar pattern and a frequency range of 20 Hz to 20 kHz. The A1 capsule is a reproduction of the CK12 capsule used in vintage microphones such as the C12 and ELAM. It has an edge-terminated, dual backplate and delivers a bright, open sound. It has sensitivity of −33 dB, a maximum sound pressure level (SPL) of 136 dB, and a signal-to-noise ratio (SNR) of 76 dB.
The A2 capsule is based on the K67 capsules from the U67 vintage micro-phone. It uses a center-terminated, dual backplate to produce a punchier, mid-focused sound. This capsule delivers a sensitivity of−36 dB, a maximum SPL of 140 dB, and an SNR of 73 dB.
The A3 capsule is a reproduction of the K47 capsule from the vintage U47 microphone. It uses a single backplate that is center-terminated and captures a full, warm, round recording of the source material. It has a sensitivity of−36 dB, maximum SPL of 138 dB, and SNR of 73 dB.
The microphone comes with a power supply, 7-pin XLR cable, power cable, wooden microphone box, wooden capsule box, shock mount, windscreen, and aluminum flight case.
The AL-Y56 is listed for US$ 999. Contact: Luke Audio, PO Box 287, Nashville, Tennessee 37070, USA; e-mail [email protected]; Web www.lukeaudiollc.com.
Vanguard V1 Multi-Capsule
Pencil Condenser Microphone
The Vanguard V1 is an FET pencil condenser microphone that comes with four interchangeable 22-mm capsules that provide cardioid, wide cardioid, hypercardioid, and omnidi-rectional polar patterns (see Figure 4). The microphone uses transformer-balanced solid-state circuits,
ultra-Figure 4. The Vanguard V1 condenser microphone.
low-noise precision-biased JFET, a custom-wound transformer, premium capacitors, and cryogenically treated components. The capsules have ultra-sensitive 4-micron thin Mylar diaphragms and precision-machined backplates. The diaphragms are lo-cated close to the front of the capsule housing to prevent comb filtering.
There are four versions of the V1 microphone kit available: a mono-phonic version, a matched stereo pair with stereo bar, a monophonic model with an additional V34C large-diaphragm capsule, and a matched stereo pair with the additional large-diaphragm capsule.
The multi-pattern V34C capsule has two 34-mm edge-terminated,
gold-sputter 3-micron thin Mylar diaphragms. When placed on the V1 microphone, its head can rotate 360 degrees for easy adjustment of positioning.
Machined brass, aluminum, steel, and zinc is used in the construction of the microphone, along with a polished nickel trim. The V1 comes with a pencil shockmount made from heavy-duty zinc, with high-impact ABS, aerospace-grade suspension, and a windowed design that allows access to microphone switches. The microphone is packaged in a pine storage case with individual carved capsule pockets and foam protection.
Pricing for the V1 kits starts at US$ 349 and extend up to US$ 1,199 for the large-diaphragm stereo matched pair. Contact: Vanguard Audio Labs; e-mail [email protected]. Web www.vanguardaudiolabs.com.
HEAR360 8ball Omni-Binaural
Microphone
The 8ball from HEAR360 is a bin-aural microphone that uses omni-directional capsules to record live, immersive, head-trackable audio for 360-degree video and virtual reality applications. It can output audio for use with Samsung Gear VR, Unity, Oculus, Vive, Facebook, and YouTube.
The microphone outputs eight channels of audio on four 5-pin mini-XLR outputs, with each output carrying two balanced audio signals. Channels 1 and 2 give the front binaural perspective, channels 3 and 4 the left, channels 5 and 6 the rear, and channels 7 and 8, the right perspective. A custom-designed cable breaks into eight standard balanced XLR jacks to allow the microphone to be connected to multi-channel audio recorders. All input channels need to have 48-V phantom power enabled.
Recordings are saved to a folder with separate monophonic files for each channel.
Plug-ins are provided for process-ing the eight-channel audio track from within the user’s DAW of choice. H360compass is a calibration tool for aligning audio with video. H360earth and H360sky can be used to spatialize audio recorded using other microphones and to blend these with 8ball recordings. H360encode is a convertor for various formats, including Facebook, YouTube, and ACN and FuMa format Ambison-ics. Samsung Gear VR is natively supported by 8ball, while a plug-in is provided for Unity. A Web player SDK for embedding 360-degree video with spatial audio to Web sites is also included with purchase.
The 8ball microphone has a fre-quency response of 20 Hz to 20 kHz and its maximum SPL is 130 dB (A-weighted). It requires 48-V phantom power. It comes with a self-centering mount that attaches the microphone beneath the camera, keeping it out-side of the camera’s field of view. The microphone centers on the view point of the camera for accuracy.
The 8ball is listed for US$ 2,500. Contact: HEAR360, 2050 Granville Ave, Los Angeles, California 90025, USA; e-mail [email protected]; Web www.hear360.io.
Rode Microphones VideoMic
Me-L
Rode Microphones has released a new directional microphone for use with iPhone and iPad. The VideoMic Me-L is a condenser microphone with a
1/
2-in capsule. It has a cardioid polar
pattern and features cancellation ports on the side that cancel sound from the rear. It connects directly to the Lightning connector of the mobile device, bypassing the audio
hardware of the phone. It has a mini-jack headphone output for recording monitoring and playback. The microphone can be pointed backwards or forwards and it comes with an adjustable mount that ensures that it is securely connected to the device. It is powered by the device it is connected to.
The microphone has a frequency response of 20 Hz to 20 kHz. Its SNR is 74.5 dB and it offers a maximum SPL of 115 dB (at 1 kHz and 1m). Sample rates of 44.1 kHz and 48 kHz are available, with a bit depth of 24.
A synthetic fur windshield is included with the purchase. The microphone is black. It weighs 28 g and measures 73.5× 20.2 × 25.7 mm. It is compatible with iOS devices using iOS 11 and higher.
The VideoMic Me-L is listed for US$ 79. Contact: Rode Micro-phones, 107 Carnarvon Street, Sil-verwater, New South Wales 2128, Australia, or Rode Microphones LLC (USA), PO Box 91028, Long Beach, California 90809-1028, USA; Web www.rode.com.
Tascam DR-10X Compact PCM
Recorder
Tascam’s DR-10X is a compact, light recorder that is designed to connect directly to the XLR output of a dynamic or battery-powered condenser microphone (see Figure 5). It measures just 5.2× 9.4 × 2.8 cm and weighs 56.3 g, making it easy to carry. It records linear PCM audio at a 48-kHz sample rate with 24-bit depth. The original recorder used broadcast WAV format but a recent firmware update now provides an option to record to MP3 also. The recorder uses microSD and SDHC cards and has a micro USB-B port for transferring audio to computer.
Figure 5. Tascam’s DR-10X audio recorder.
The DR-10X has both manual and automatic gain settings, a low cut filter, and a limiter. It features a dual recording mode that captures the main audio but also a backup copy at a lower level in case any unexpected issues with distortion arise. A sliding record button with a hold function is used to prevent accidental switching on/off of the recording. The slide switch can be held down while switching on the recorder to commence recording immediately upon start-up.
The recorder requires just one AAA battery, which delivers 10 hours of recording. The audio file will automatically close if the battery runs out, preventing loss of audio. The recorder has an electronic display that shows the battery level. There is a built-in clock and the broadcast WAV files that use time data are supported. New files can be automatically
Figure 6. The 2X2M audio interface from M-Audio.
created at regular time intervals during recording. There is a built-in mbuilt-ini-jack headphone output for monitoring recording. The latest firmware update also adds real-time level meters for recording and a new level setting in the dual recording mode (−12 dB and −6 dB are now available).
The DR-10X is listed for US$ 119.99. Contact: TEAC America, 7733 Telegraph Rd., Montebello, Cal-ifornia 90640, USA; Web tascam.com.
M-Audio M-Track C-Series Audio
Interfaces
The new 2X2 and 2X2M USB-powered interfaces from M-Audio are 192 kHz,
32-bit interfaces that include the company’s new Crystal preamplifiers and+48-V phantom power.
The 2X2 model has a balanced combination XLR/1/
4-in input and
a 1/
4-in instrument input with a
custom-designed gain and impedance stage for direct connection of guitar and bass. The 2X2M version (see Figure 6) has two of each input and an additional MIDI input/output on 5-pin ports. Both interfaces have
1/
4-in stereo outputs, and a full size
headphone output with independent level control.
The control panel features a large machined aluminum volume knob in the center. High speed, zero latency monitoring is available and there is a USB/Direct balance knob.
The frequency response on all inputs is 20 Hz to 20 kHz. The gain range is 55 dB on the microphone and line inputs and 24 dB on the instrument inputs. The interfaces have a SNR of 104, 101, and 98 dB on the microphone, line, and instru-ment inputs, respectively. The main outputs have a frequency response of 20 Hz to 20 kHz, SNR of 106 dB, total harmonic distortion with noise of 0.0005 percent, and a maximum output level of+13 dBu (at 1 kHz and −1 dBFS).
The interfaces can connect to computer (Macintosh or Windows) using a standard USB port or a USB-C port, through which they are also powered. They have a metal chassis, weigh 2 lbs and measure 6× 2.8 × 7.8 in. Purchase includes a set of
software applications: Steinberg’s Cubase LE, AIR Music Technology’s C-Series Virtual Instrument Suite, and AIR’s Creative FX Collection.
The M-Track 2X2 is listed for US$ 99 and the 2X2M for US$ 149. Con-tact: World Headquarters, 200 Scenic View Drive, Cumberland, Rhode Island 02864, USA; Web www.m -audio.com.
Audient iD44 Audio Interface
The iD44 audio interface from Au-dient is a 20-input, 24-output audio interface with four Class-A console microphone preamplifiers and 60 dB microphone gain. The interface sup-ports sample rates up to 96 kHz, at 24-bit sample depth. It features two discrete JFET direct inputs for instruments, which replicate the in-put stage of a classic valve amplifier. There are two balanced inserts and the insert returns have direct access to the A-D convertors. Two ADAT inputs/outputs facilitate expansion of the interface by a further 16 channels. The interface has two independent stereo headphone outputs, along with main and alternative speaker outputs.
The iD44 has console-style moni-tor controls with dedicated talkback function, cut and dim, polarity, mono, speaker select, and cue mix monitor-ing. Three function buttons can be programmed for monitoring func-tions. The talkback feature allows any audio source connected to a computer to be used as a talkback microphone, including built-in or USB microphones, so that the high-resolution microphone preamplifiers can be used for recording.
A ScrollControl function allows the volume encoder to double as a scroll wheel to control DAW and plug-in parameters. The interface
connects to Macintosh and Windows machines via USB 2.0, and a standard USB and USB-C cable are included with purchase. The interface requires an external 12-V DC power adapter and cannot be USB bus-powered. A Word clock output is provided.
The interface has a low noise floor. It offers a dynamic range of 126 dB on the D-A convertors and the headphone output. It has a total harmonic distortion with noise of just 0.0007 percent. It has an all-metal body with hand-milled aluminum knobs. The interface measures 27.5× 5.1× 17.4 cm and weighs 1.75 kg.
It comes with iD Mixer, a mixing application that allows the user to create up to four different stereo cue mixes, with customization and nam-ing for channels, advanced routnam-ing options, and the ability to assign functions to the hardware controls.
The iD44 interface is listed for US$ 699. Contact: Audient, Aspect House, Herriard, Hampshire, RG25 2PN, UK; Web www.audient.com.
CEntrance MixerFace R4
The MixerFace R4 is a recording audio interface for mobile phones, tablets, and computers. It is compatible with Android, iOS, Macintosh, and Windows platforms.
The interface has two combination Neutrik inputs for XLR and line-level inputs, with a Hi-Z switch to enable direct input from guitars and basses. The microphone inputs have Jasmine microphone preamplifiers and can supply 48-V phantom power, if required. An auxiliary stereo input is also included for mixing-in external stereo sources. The interface connects to smartphone, tablet, and computer via USB. There are balanced outputs, an unbalanced stereo output, and a headphone output.
Top panel controls on the interface include channel gain knobs with peak and signal LEDS, monitor blend knobs, an auxiliary stereo input level adjustment, a headphone level, and a monitor level. A gain switch provides 20 dB of gain reduction. A 100-Hz high-pass filter with 3 dB per octave roll off is also available on the inputs.
The MixerFace R4 supports sample rates from 44.1 kHz to 192 kHz, with 16-, 24-, and 32-bit sample depth. Its frequency response is 20 Hz to 20 kHz. The SNR is−113 dBu and the total harmonic distortion with noise is reported to be just 0.002 percent. The interface also has two custom 10 ppm precision clocks with 1 ps of jitter.
A long-life rechargeable lithium-polymer battery allows the interface 8 hours of operation and a battery level indicator is located on the bottom panel. The device can be charged with a standard USB charger through a USB microB cable. The chassis is made of black aluminum that has been hard-anodized to make it scratch resistance. The interface measures 4.76× 2.76 × 1.42 in and weighs 4.3 oz. It ships with a felt carry pouch and a USB cable.
The MixerFace R4 is listed for US$ 349.99. A protective carry case is available as an additional purchase at US$ 19.95. Contact: CEntrance, Inc., 8817 Mango Ave., Morton Grove, Illinois 60053, USA; e-mail [email protected]; Web www.centrance.com.
Behringer X-Touch One Control
Surface
Behringer’s X-Touch One is a Uni-versal Control Surface for use in live performance or home studio setups (see Figure 7). It incorporates a
Figure 7. Behringer’s X-Touch One control surface.
touch-sensitive motorized fader, an LCD scribble strip, rotary encoder, 34 illuminated buttons, an eight-segment LED meter, a jog/shuttle wheel, and dedicated transport con-trols.
The scribble strip at the top of the main panel shows the name of the active track and the parameters
being edited. The touch-sensitive 100-mm fader is motorized and fully automated. There is a rotary encoder with an LED collar on the very top left of the control panel for easy editing of parameter settings, such as pan. There are 34 separate illuminated buttons for quick and easy access to a range of DAW functions.
The X-Touch One supports HUI, Mackie Control, and MIDI proto-cols for communication with audio and music software applications. The functions for use with Mackie Control are labelled on the control surface itself but a user mode for assigning custom functions to the controls is also available. In addi-tion, a set of assignment presets for popular DAWs (including Pro Tools, Logic, Studio One, Reaper, and Waveform), are included, along with overlay templates with control labels.
The rear panel has a footswitch port for connecting external remote control devices. It also features a USB hub with two Type-A ports for con-necting multiple controllers together, and a Type-B port for connection to computer. The control surface comes with a 9-V DC power supply.
The X-Touch One is listed for USS$ 249.99. Contact: Behringer; Web www.musictri.be/brand/behringer/.
Presonus FaderPort
Presonus’ FaderPort is a new control surface for use with DAWs (see Figure 8). It features 24 buttons that give access to 40 functions using a shift key, a large rotary encoder, and a 100-mm touch-sensitive, motorized fader, which can be useful for cre-ating fades and automation in real time.
The center panel of the FaderPort features a set of Session Navigator controls. There include the rotary encoder, a set of eight buttons for mixing and editing functions (link, pan, channel, scroll, master, click, section, and marker), and previous and next arrow buttons. The user can easily scroll through the timeline, zoom in and out of audio files, add and navigate markers, and set
Figure 8. The Presonus X-Touch One control surface.
the main level using these controls. Below this is the transport section, with buttons for play/pause, stop, fast forward, review, record, and loop. Further channel-level buttons are provided for solo, mute, arm/arm all, bypass/bypass all functions. Touch, write, latch, trim, read, and off but-tons are also included for automation control.
The control surface supports Mackie Control and HUI protocols for use with a range of both Macin-tosh and Windows DAWs, including Logic Pro X, Cubase, Pro Tools, and Presonus’ Studio One. The FaderPort connects to computer using a USB 2.0 port. A footswitch port is included for hands-free operation. It requires a 12-V DC external power supply. The FaderPort measures 1.8× 5.5 × 9 in and weighs 1.5 lb. The Studio One DAW Artist edition is packaged with the device.
The FaderPort is listed for US$ 299.95. Contact: PreSonus Audio Electronics, 18011 Grand Bay Court, Baton Rouge, Louisiana 70809, USA; Web www.presonus.com.
