Tuesday, 13 March 2012 Written by Tim Goodyer
Monday, 17 February 2014 Written by Tim Goodyer
TC Electronic has released a software update for its DB6 intelligent loudness management system.
Using the new software, broadcasters are able to recall up to 16 Scene Presets by using GPI 1 and 2 together. DB6 offers a variety of ways to engage or disengage processing automatically, depending on format, level and physical inputs, while remote control can be achieved via Ethernet, GPI and SDI metadata.
The GPI implementation of DB6 is extensive, offering switching between up to 16 Scene Presets using only two of its four GPI inputs. Based on this, DB6 may be controlled during playout, or a simple preset selection panel can be designed with easy access to presets and/or emergency override.
Further, it is now possible to control DB6 by multiple instances of the Icon remote, adding a new level of operational flexibility to the platform. It is also possible to copy entire User preset Banks to or from files stored on a PC or Mac.
Wednesday, 24 October 2012 Written by Tim Goodyer
Algorithms to Measure Audio Programme Loudness and True-Peak Audio Level
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Monday, 05 November 2012 Written by Tim Goodyer
Building commercial and technical support for its growing base and realisation of our expansion plans, Netherlands-based loudspeaker manufacturing specialist Alcons Audio is invinting applications for the following positions:
Audio DSP developer
Sales Support Engineer, Germany
More information and application details are available from Alcons Audio.
Tuesday, 17 December 2013 Written by Tim Goodyer
Audio Precision has announced an Android app for testing smartphone audio quality.
Serving as an interface between the Android device and one of Audio Precision’s APx audio analysers, the app allows fast, automated profiling of new phones and tablets for evaluation of the audio performance of different models and manufacturers against a common standard.
· Route audio through various analogue signal paths not normally allowed by stock Android OS (for example, making a closed-loop test from analyser to analogue line-in to device’s speaker and back to analyser)
· Use device’s record feature, then transfer the resulting file to an APx audio analyser over USB for analysis
· Test Bluetooth signal paths, including the playback of stimulus waveforms over Bluetooth Handsfree Profile (HFP) without initiating a phone call over the network
· Automate the process with the APx desktop application to control the device over USB, allowing audio test sequences to run through all available signal paths with no user interaction required.
The ability to route analogue audio through arbitrary inputs and outputs in loopback mode makes testing much faster than record and playback and enables real-time electroacoustic measurements. In addition, the default media player is bypassed, so there is no risk of user settings, EQ or resampling interfering with measurements.
The ability to open a SCO and send signals over Bluetooth HFP directly from the phone takes the mobile network’s audio performance out of the equation. The stock Android OS does not allow files to be streamed over HFP. The AP app overrides this behavior, allowing direct evaluation of the Bluetooth connection quality independent of the mobile network (the weakest link in the signal chain).
The AP Smartphone App for Android supports sine, continuous sweep and multitone signals for music playback signal paths. For voice band, AP recommends using PESQ or POLQA MOS (mean opinion score) perceptual audio measurements, which are fully supported by the AP smartphone app and are incorporated in the accompanying APx project files and test waveforms. The app supports all APx supported profiles and codecs: A2DP (with SBC and APTx codes) and HFP/HSP (with CVSD and mSBC wideband voice codecs).
The new desktop app and accompanying APx project files allow all the testing to be automated, significantly reducing test time, and minimizing the opportunity for operator error.
The APx Smartphone app is available for download at the AP.com website at no charge as part of Technote 120. The app requires Android v4.0 and higher and a connected APx audio analyser to function.More: http://ap.com
Thursday, 23 January 2014 Written by Tim Goodyer
Audinate has released Dante Controller v3.5, an update to its free networking management configuration and monitoring application.
New features add advanced real-time network monitoring functionality, including multicast bandwidth use, clock health monitoring and latency management statistical information. These provide quickly identification and rectification of potential issues on a network, while preserving Dante Controller’s ease-of-use.
‘Many of the new features in Dante Controller 3.5 were implemented to directly meet specific customer requirements’, says Audinate VP of Product Management, Gary Southwell. ‘Dante Controller 3.5 includes many features that AV and IT professionals could only dream of a few short years ago.’
Enhanced Latency Reporting
· New histogram-based reporting and tracking of transmitter latencies – provides at-a-glance information about the latency performance of audio flows for supported devices.
· Latency histograms indicate if latency play-out settings are properly configured, and also provide an overall view of the health of the network: allowing an operator to quickly see if an audio flow has been properly configured and if underlying network issues could potentially develop into an audio quality problem.
Improved Clock Health Monitoring
· Clock health is now tracked on a per-device basis via dedicated performance histograms. This enables the quick and easy detection of difficult-to-diagnose network issues, which are impacting clock accuracy - such as clock traffic bottlenecks, and low-performance external word clocks.
New Preset Tool Suite
· New preset system exposes XML-based network configuration presets for use in off-line configuration tools. This simplifies the setting up of new networks, and the reconfiguration of existing networks.
· Presets enable simpler switching between standard configurations and special event configurations. Provides greater confidence in the ability to share audio network configurations for multiple uses and/or user groups.
Redundant Network Connection
· Enables visibility of primary and secondary networks at all times. Redundant devices can still be seen and managed via the secondary network in the event of a problem with primary network connectivity.
· Simplifies redundant network set-up, and allows verification of network configuration.
· Provides real-time notifications about connectivity status on both primary and secondary networks.
New Channel Grouping Capability
· Allows simple viewing and configuring of large numbers of device channels, in groups of 16.
· Each group can be expanded then collapsed when set-up is complete.
Improved Network and Device Status Information
· Network and device information has been enhanced and restructured across two tabs (including a new Network Status tab) to provide a clearer and more useful overall picture of the network.
· New notification icons provide real-time information about a range of network and device events.
Monday, 08 April 2013 Written by Tim Goodyer
Calrec Networking Primer
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Wednesday, 27 March 2013 Written by Tim Goodyer
Tascam has aimed its DR-60D at on-set filmmakers and videographers using DSLR cameras.
The DR-60D is a four-track solid-state recorder that is designed fit under any camera or into any rig, and record up to 96kHz, 24-bit audio in WAV/BWF formats to SD/SDHC media. The unit will handle simultaneouns four-track recording, includes M-S decoding, and has a slate for marking tracks, It also offers RC-10 Remote compatibility. The body structure, soft-touch keys and dials help to eliminate handling noise.
· Tascam HDDA microphone preamps
· Independent recording level adjustment
· Two XLR/TRS inputs
· +4dBu line level input
· Phantom power (24V/48V)
· High-output mic input on Input 3-4
· Camera out connector for DR-60D mixer
· Camera in connector for sound monitoring
· Independent line out and headphone output
· 50mW/ch headphone output
· Hold switch to prevent accidental operation
· Quick button for easy access to various functions
· 128x64 pixel LCD with backlight
· USB 2.0 audio transfer (Mini USB cable included)
· Internal mixer: pan and level controls
· Low-cut filter(40/80/120Hz)
· Limiter (1/L and 2/R can be selected for link-operation)
· Delay compensation for mic position (±150ms)
· Slate tone generator (auto/manual)
· Selectable duration of slate tone from four positions (0.5s/1s/2s/3s)
· Selectable slate tone generator (off/head/head+tail)
· File name format can be set to use a user-defined word or date
· Dual recording function allows two files to be recorded simultaneously at different levels
· Auto-record can automatically start and stop recording at set level
· Pre-recording (2s sound buffer before recording)
· Self timer for solo recording
· New file starts recording automatically without interruption when maximum file size is reached
· Track incrementing
· Jump back and play function
· Equalisers function for playback and level alignment
· Resume function to memorise the playback position before the unit is turned off
· Mark function up to 99 points per audio track
· Divide function
· Tripod mounting threads (bottom) and DSLR screw attachment (top)
· Dedicated jack for wired RC-10 remote control or RC-3F footswitch (sold separately)
· Operates on 4 AA batteries, an AC adapter (sold separately) or USB bus power
· Can extend battery life with BP-6AA battery pack (sold separately)
The DR-60D is expected to have a street price of US$399.
Monday, 02 December 2013 Written by Tim Goodyer
Sonifex has added a new audio distribution amplifier to its Redbox audio range.
‘We designed this product for a niche application of distributing a mono or stereo audio feed to multiple active speakers for an emergency audio distribution system,’ says Sonifex Sales Manager, Richard Butlin.
The RB-DA24MD was designed in response to a particular request before becoming a ‘standard’ Redbox: ‘The customer has a number of office desk clusters, each with a small speaker mounted underneath them, so that in the event of an emergency, everyone in the building gets a clear emergency signal,’ Butlin explains. ‘Because it has so many outputs, the system can cover 100 desks with only five units.’
The RB-DA24MD is a 24-way amplifier, with two inputs that can be individually routed to 12 outputs or mixed and routed to all 24 outputs. The inputs can be configured as either dual mono, Input 1 routed to Outputs 1-12 and Input 2 routed to Outputs 13-24, or mixed-mono, Inputs 1 & 2 mixed at a preset level and routed to all 24 outputs.
The XLR inputs and D-type outputs are electronically balanced and can be wired unbalanced. Each output is individually buffered so that a short circuit on one won’t affect the others. Each output is also protected against connection to both POE (power over Ethernet) and phantom power circuits.
The RB-DA24MD has master gain controls for Inputs 1 & input 2, which are preset potentiometers accessible through the rear panel. These controls allow the gain to be adjusted from -15dB to +15dB, useful for normalising consumer to professional signals and vice versa. An LED power indicator on the front panel shows that the unit is powered. Also a 125Hz 6dB/octave roll-off filter can be switched in.
Friday, 21 December 2012 Written by Tim Goodyer
After its presentation at NAB and IBC, Lawo’s first video device V__pro8 has been brought to v1.0 level released as a regular product. Using a GUI based on HTML5, it combines a number of operations for maximum efficiency and operating safety at optimal cost.
Lawo has developed the V__pro8 for displaying video in a small footprint, connecting signals of different video formats as well as providing a bridge between audio and video. It offers high-quality cross-format conversion, colour correction as well as embedding and de-embedding. Modern high-density audio connections, such as Ravenna and Madi, provide video-to-audio bridges within the studio infrastructure.
The V__pro8 also works as a flexible matrix system: thanks to a 8 x 8 video matrix and a 384 x 384 audio matrix it is possible to switch any signal to any other signal.
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