RB-DS2 Stereo Delay Synchroniser & Time-Zone Delay
The RB-DS2 is a stereo audio delay synchroniser used for resynchronising audio to video following delay processes such as standards conversion, transmission delay, logo insertion, video aspect ratio conversion and network delays. It can be used for fixed installations to correct a permanent audio delay, or on an intermittent basis to provide occasional correction, for example for live links. Accepting digital audio signals up to 96kHz, 24 bit, the sonic quality of the RB-DS2 is superb and silent switching is used to provide the smoothest, cleanest audio delay available.
The RB-DS2 has both balanced analogue and AES/EBU digital audio inputs and outputs on 3 pin XLR connectors. It can act as a combined A/D and D/A unit meaning that analogue inputs can be delayed and output as AES/EBU or vice-versa. It is a stereo delay, but can also be used as a dual mono delay, to process each audio path separately, or as a mono delay only.
As standard the RB-DS2 can provide up to 10.5 seconds of delay at 96kHz sampling, 24 bit (42 secs at 48kHz, 16 bit). An internal Compact Flash™ expansion allows up to 2GB of memory to be accessed providing delay times of over 4 hours, for example, to delay a programme output across different time-zones, or to shift a broadcast programme by 1 hour for a satellite rebroadcast. Delay times can be selected in samples, fields, frames, metres, milliseconds and with the Compact Flash™ expansion, in hh:mm:ss. Frame and field definitions can be for PAL (25 frame) or NTSC (30 frame) signals.
A front panel blue vacuum fluorescent display with rotary controller is used for selecting the various settings of the delay, which include the source (analogue or digital), channels, sample rate, sample bit width, format (PAL or NTSC), delay units and the delay itself. Additionally, input peak digits can be selected from +12dBu, +18dBu and +24dBu for FSD and two left and right pre-set potentiometers on the rear panel allow the input gain range to be altered by ±3dB around the selected peak digits.
The analogue output gain range can be altered in software from -6dBu to +24dBu output level, ref FSD. Both analogue and digital outputs can be separately muted and a front panel Bypass button disengages electro-mechanical relays to divert both analogue and digital inputs to their outputs. This is also disengaged automatically when a power-fail occurs.
All of the settings in the unit can be saved to one of 8 configuration settings. These Configs can be viewed, edited, saved and loaded, and also remotely loaded by using one of the 8 GPI contacts, meaning that any setting, such as delay time or Bypass, can be altered instantaneously using a GPI signal. The RB-DS2 also has an RS232 serial port for remotely controlling the unit and there are 4 remote outputs which can be used for signalling. The front panel controls can be locked-out for situations where remote control is being used to run the unit, or where physical security is required.
Housed in a red-anodised aluminium rack-mounting case, the delay is intended to be mounted in a studio or equipment room and operated using the controls on the front panel, or the various remote options. The controls on the front panel can be locked out, in situations where remote control is being used to run the unit, or where physical security is required. Sonifex SCi software, available free of charge as a download from Sonifex, when installed on a suitable PC, provides graphical access to all remote control and configuration options via both RS232 and USB interfaces. Alternatively, commands may be issued from any text-based terminal application.
The front panel is dominated by a vacuum fluorescent display which displays permanently the current delay value, in seconds, and the current status of the unit. An associated rotary selector with integral push-switch is used for selecting the various settings, which include the start delay and dump modes, safe period, source (analogue or digital), sample-rate and sample bit-width as well as the required delay time. Four push buttons offer ‘Build Delay’, ‘Exit Delay’, which ramps the delay down to return to real-time output, ‘Audio Bypass’, which disengages relays to divert both analogue and digital inputs to their outputs and ‘Cough’, which allows locally generated sounds being presented at the inputs, such as the presenter coughing or equipment switching noises, to be discarded. A larger button operates the dump function, which has two different modes. The first removes a section of audio that has already been buffered, by a pre-selected amount. The second mode plays a pre-selected audio file on the memory card inserted in the CF slot, which supports cards up to 2GB. When the file has finished playing, the delay is then equal to the duration of the file. The dump button can be used multiple times to use up the built-up delay and once used, the unit starts to rebuild the original delay time automatically. As a last resort, all the buffered audio can be discarded by pressing and holding the dump button, thus activating the ‘Drop’ function. A dedicated record mode allows audio presented at either the analogue or digital inputs to be recorded to a linear WAV file on the memory card. Files may be transferred to and from the card using a PC.
The rear panel has two pairs of XLR connectors for balanced line-level analogue inputs and outputs. The inputs have adjacent trimmers of level. A further pair of XLRs are used for the AES/EBU digital input and output. A 15-pin male D-Sub connector provides external control; eight inputs and six outputs are assignable freely. The inputs can be used to trigger any of the unit’s functions such as build delay, activate cough or enter record mode and start a new recording. The outputs can provide external signaling to indicate when certain events have occurred such as the delay reaching the required value or the outputs being muted. A 9-pin male D-Sub connector provides an RS232 serial port. Power is connected by a filtered IEC inlet, with adjacent fuse.
Sampling frequency: 32kHz, 44.1kHz or 48kHz. Sample width: 16-bit or 24-bit. Minimum delay: 2 seconds. Maximum delay: 55/27.5 seconds at 16/24 bits, 32kHz; 40/20 seconds at 16/24 bits, 44.1kHz; 37/18.5 seconds at 16/24 bits, 48 kHz.
Analogue audio inputs: +28dBu maximum, 10kohms. Analogue and digital level: +12dBu / +18dBu / +24dBu for FSD. Analogue trimmers: +/-3dB. Noise: Better than -101dBFS RMS, A-weighted, 24bit. Dynamic range: Better than 110dB. Distortion: Better than 96dB THD plus noise, 1kHz. Analogue audio outputs: +24dBu maximum, 50 ohms. Dynamic range: 100dB minimum. Gain range: -6dBu to +24dBu output level, ref FSD. Power source: 85-264V AC, 47-63Hz, 60/30W peak/average. Dimensions: 480 x 158 x 43 (w x d x h)mm. 1U rack mounting. Weight: 1.7kg.
Included accessories: Handbook, AC power cordset with plug.
Optional accessories: Compact Flash memory card up to 2GB capacity, PIO type 4 or higher.
ACOUSTIC ECHO CANCELLER
The RB-AEC is primarily designed for the benefit of studio personnel in television and radio. When a studio presenter’s microphone signal is played out through a monitor speaker in the control room, it can be picked up by the control room microphone(s) and returned to the presenter’s earpiece as an undesirable echo. This unit ‘learns’ the environment and can be used to remove the echo from the earpiece, using digital processing.
In circumstances where green-screen video processing is taking place, the delay can be greater than 200ms. Additionally, the dimensions, occupancy and distance between mouth and microphone can further influence the echo. The RB-AEC is used to remove the entire output of the control-room monitor-speaker from the presenter’s feed, by adapting to the environment in which the control-room microphones are placed. Although acoustic echo cancellation is more commonly implemented in telephony systems, the Sonifex RB-AEC is designed to produce broadcast quality cancellation.
The post-processed transmission output program from the studio (A) is sent to the RB-AEC as an analogue or digital audio signal (the stereo input is auto-sensing) which ‘acts as a mix-minus to the input signal (B) from the control room’. The RB-AEC removes the unwanted acoustic echoes so that the audio sent to the presenter’s earpiece (C) is free of echoes and reflection artifacts.
The device may also be used during a ‘phone-in’ to remove delayed caller audio from the telephone line.
The unit can be controlled remotely over an Ethernet connection using the built-in web server. By default the unit is configured for dynamic addressing using DHCP and ‘Auto-IP’. If a static IP address is required then this must be configured through the web server.
The device is packaged in the familiar Sonifex Redbox rack-mounting chassis, IU high. The front panel has an LED to indicate power.
The rear panel has two 3-pin, female, XLR connectors for the line level analogue or digital inputs for the ‘far end’ and ‘near end’. Three 3-pin, male, XLR connectors provide analogue outputs one and two as well as the digital output. A bank of four miniature (DIL) switches adjust settings, a 9-pin, female, D-sub connector provides GPIO connections and an RJ45 connector is provided for Ethernet. Power is connected via an IEC inlet, with an adjacent fuse.
Audio inputs: +18dBu / 0dFS maximum, 20k ohms / 110 ohms with termination set, analogue / digital. Input gain: 0, +6, +12 or +18dB digital gain, switchable. Distortion: 0.02%, 1kHz, +8dBu output. Noise: -84dB RMS, unity gain, ref +8dBu output. Response: 20Hz-12kHz +0/-0.5dB. Rejection ratio: Typically 20dB input to output on complex waveforms, reference peak level of 0dB. Audio outputs: +18dBu / 0dFS maximum, 50 ohms / 110 ohms, analogue / digital. Output sample rates: 32kHz – 192kHz, selectable. Power source: 85-264V AC, 47-63Hz, 10W maximum. Dimensions: 480 x 108 x 42 (w x d x h). Weight: 1.5kg.
Included accessories: Manual, AC power cable.
Optional accessories: RB-RK3 Rear rack panel kit.