| Audio signal processing, routing and infrastructure |
Matrox DSX Core
Active Optical Cables (AOC) for Video
Active Optical Cables (AOC) for Video
• Does not compress or alter the video in any way
• No power supplies - installs in small and tight spaces
• Provides optimum video quality for precise visualization
• Customizable cable lengths cover exact distance requirements
• Eliminates signal delays transporting video signals in real-time with no change or color reduction from image
• Transports video signals over fiber, offering same interference immunity for optimum security
• Costs less compared to other AV extenders
V__matrix vm_avp SDI-to-IP Gateway
V__matrix vm_avp SDI-to-IP Gateway
In base form, vm_avp provides encapsulation and de-encapsulation of 3G, HD and SD-SDI (ST2022-6/7 only) to IP ST2022-6 and ST2110-20/21/30/31/40, making vm_avp is the logical choice for both gateway and purely IP-based A/V processing, providing up to 160 SDIIP conversions in 3RU. ST2022-7 seamless protection switching is standard, with IP stream format conversion and frame accurate video switching using destination-timed clean and quiet switching (MBB & BBM) with audio V-fade during switching.
To further expand the usefulness of vm_avp, several optional add-on licenses are available:
- +audio – adds embedding/de-embedding and shuffling of audio from both IP and baseband I/O with sample rate conversion. 40 TX and RX instances of RAVENNA/AES67/ST2110-30/-31 streaming and an audio crossbar of 512 x 512.
- +audio_matrix – an additional upgrade to the +audio option, the +audio_matrix option provides a total of 88RX and 128TX instances of RAVENNA/AES67/ST2110-30/-31 streaming and an increased audio crossbar of up to 5,312 x 5,312.
- +madi – enables use of V__matrix BNC inputs and outputs to interface with MADI signals (AES10, 48kHz, 64 channels).
- +fs – adds framesync, frame phaser, sample rate conversion and audio/video delay functionality for both IP and baseband video inputs; also provides Dolby-E® auto-alignment functionality.
- +vc2 – adds visually lossless VC2/DiracPro ultra-low latency encoding and decoding; ST2042 low-delay profile.
- +proc_cc – adds YUV & RGB color correction, test pattern generator/inserter, and test tone generator.
- +12g – Adds support for 12 Gbps video standards and provides cross conversion capabilities between UHDTV1 Single-Link and Quad-Link.
V__matrix vm_udx 4K/HDR Format Converter
V__matrix vm_udx 4K/HDR Format Converter
The vm_udx app provides a format conversion engine capable of processing four SD, HD, 3G or one UHD path for IP and/or SDI signals. Each path provides audio embedding/de-embedding/shuffling functionality. Audio gain, delay and sample rate conversion can be accessed through independent processing blocks, which can be inserted at any point of the processing chain. Eight instances of broadcast quality RGB and YUV color correction and video proc are also available as processing blocks for use by any video source, whether SDI or IP, and available both pre- and post-format conversion.
For even more functionality, add the +HDR option to enable four instances of SDRHDR color space conversion using 3D lookup tables. A large selection of LUTs especially developed for live production are included, and custom LUTs can also be uploaded too. The included LUTs allow for conversion between SDR and HDR in HLG and PQ.
RƎLAY Virtual Radio Software
RƎLAY Virtual Radio Software
RƎLAY virtual radio software from Lawo brings professional radio production to your PC. Mixing, streaming, dynamics processing, routing and even AoIP signal monitoring, using standard RAVENNA / AES67 networking via Lawo’s virtual sound-card software for Windows. Mix and route any audio produced by PC applications, streams from your AES67 network, or any audio signal your PC can ingest. All this power at your fingertips, using touch-screen GUIs so intuitive that talent can learn them in minutes. No exotic hardware needed: just install RƎLAY software on your PC or laptop, and go to work. Perfect for production, live studios, newsrooms — even remotes and OB work.
There are four RƎLAY Virtual Radio apps:
- VRX Virtual Radio Mixer, a powerful mixing console that runs in a virtualized environment. No racks of gear to devour space and money – RƎLAY VRX runs on standard PCs and laptops, with an intuitive, multitouch-enabled interface that’s easy to learn and use. Available in 8-fader and 4-fader versions.
- VPB Virtual Patch Bay, a cross-point patch bay on your PC. RƎLAY VPB gives you unrestricted control over the routing and mixing of audio signals on your computer. Patch inputs to outputs, combine multiple channels, even apply processing using the included Lawo Processing Suite, or your favorite VST plug-ins.
- RƎLAY Virtual Sound Card, an 8×8 AoIP driver (with up to 64 bi-directional channels of stereo I/O) for Windows. RƎLAY VSC turns PC audio into clean, pristine AES67 streams to interface with your studio’s RAVENNA / AES67 network — without the expense of traditional sound cards.
- AoIP Stream Monitor keeps you informed of the state of your critical program streams with real-time confidence metering, LUFS meters, silence sense, audio level alerts, etc., displaying realtime informatics for up to 16 AES67-compatible audio streams. Average and peak bargraph meters for each channel are onscreen constantly, accompanied by a level readout in Loudness Units, and visual alarms for under-level, over-level and error states, plus multiple info tabs with stream health, performance history, and detailed SDP information. Perfect for Master Control displays, stream diagnostics, audio level checks, and much more.
A__line WAN-Capable Audio-over-IP Nodes
A__line WAN-Capable Audio-over-IP Nodes
Lawo A__line nodes are designed to serve as IP audio stageboxes for mc² consoles, audio extensions for the V__matrix ecosystem, or as stand-alone IP audio gateways.
A__line devices provide distributed audio connectivity that makes it easy to scale audio I/O capacity in a networked system — temporarily or permanently. SMPTE ST2110-compliant, A__line offers a granular audio format selection similar to the flexibility found in baseband modular I/O systems, but eliminates system design limitations like maximum channel count per frame, fixed channel capacity per device interconnect, or fixed overall system size. Audio endpoints connected to A__line nodes can be seamlessly shared on LAN or WAN, and thanks to their fully standardized streaming technology, can interconnect to a wide variety of IP-enabled broadcast devices — A__line nodes are brand-agnostic.
A__line devices boast exceptional audio quality. Discrete, Class-A microphone preamplifiers deliver a superb dynamic range of 119dB(A), ultra low noise at all gain levels and a perfectly flat frequency response. Versatile analog I/O accommodates levels as high as +24dBu before clipping. Insertable, precise sample rate conversion is available on each AES3 input. For multichannel baseband interfacing, A__line features bi-directional MADI access via SFP. All devices employ the ST2110-30/31 and AES67 standards to transport uncompressed audio in real-time on Layer-3 IP networks, with ST2022-7 Seamless Protection Switching with dual-redundant network interfaces and ample receive buffer capacity to meet ST2022-7 class C specs for LAN and WAN deployment. Two redundant power inlets complete the package, along with PPM metering for all Analog and AES3 interfaces and PTP/Wordclock sync and conversion.
The A__line family includes:
- A__stage48 – 32x mic/line in, 16x line out, 8x AES3 in, 8x AES3 out, 1x MADI, 8/8 GPI, 3RU
- A__stage64 – 32x mic/line in, 16x line out, 8x AES3 in, 8x AES3 out, 1x MADI, 8/8 GPI, 4RU
- A__stage80 – 32x mic/line in, 32x line out, 8x AES3 in, 8x AES3 out, 1x MADI, 8/8 GPI, 3RU
- A__mic8 – 8x mic/line in, 4x line out, 8/8 GPI, 1RU
- A__digital8 –8x AES3 in, 4x AES3 out, 8/8 GPI, 1RU
- A__digital64 – 32x AES3 in with SRC, 32x AES3 out, 1x MADI, 8/8 GPI, 3RU
- A__madi6 – 6x MADI, 1RU
When controlled with Lawo’s VSM broadcast control system, A__line stageboxes provide an advanced set of networking options; VSM manages A__line’s AoIP connections, I/O settings, its non-blocking internal routing matrix, and GPIO signals for smooth integration into an overarching operational workflow.
Power Coreᴿᴾ IP Audio I/O & DSP Node for Remote Production
Power Coreᴿᴾ IP Audio I/O & DSP Node for Remote Production
Lawo’s Power Coreᴿᴾ is a full-featured remote production solution for mc² audio consoles, providing integrated modular I/O, DSP and IP streaming capabilities, all in a compact 1RU device. It accomodates 64 channels of MADI audio via its front-panel port, and I/O count can easily be expanded using the eight rear-panel slots which can be loaded with Mic, Line, AES3 and GPIO cards, in any combination, for a total I/O capacity of up to 128 channels.
Designed for mission-critical applications, Power Coreᴿᴾ includes ST2022-7 compliance for network redundancy and Class C jitter / network latency robustness to eliminate the need for dedicated WAN gateways and minimizing the complexity of remote productions while reducing single points of failure. Its IP interface complies with ST2110-30/-31, AES67 and RAVENNA networking standards to deliver maximum interoperability within your production suite. To accommodate the most sophisticated workflows, Power Coreᴿᴾ features comprehensive audio connectivity via two redundant 1GbE SFP ports for Audio-over-IP, one MADI port (with a 2nd port for interface redundancy), and eight modular I/O slots that can accommodate mix of Mic, Line and AES3 cards. A unique studio card with Mic/Line inputs & outputs plus two headphone amplifiers is available as well.
Power Coreᴿᴾ DSP capabilities include 64 fully-featured processing channels and provide low-latency on-site monitor and IFB mixing. mc2 consoles at home have full control of all relevant channel parameters (gain, fader, mute, EQ, dynamics, Aux Send Level, etc.) of the DSP node at its remote location.
All I/O parameters of the Power Coreᴿᴾ can be remote controlled from an mc² console’s channel strips; in addition, mc² consoles can control the DSP channels inside Power CoreRP to offer control capabilities for up to 64 input channels and 16 stereo Aux busses. Remote channels can be mapped to the host console’s surface just like any other source and offer parameter control for Fader, Mute, EQ+Filters, Dynamics, Delay plus the analog and digital input section. Remote inputs and Auxes of Power Coreᴿᴾ can even be linked to local DSP channels of the host console to ensure continuous linking of parameter values.
In addition to the control integration from an mc² console, Power Coreᴿᴾ features a touch-screen optimized control GUI based on the Lawo VisTool screen design solution. This provides on-site as well as remote access to all parameters of Power Coreᴿᴾ, which is convenient for setting up prior to connecting the host console. Power Coreᴿᴾ ‘s unique remote-control possibilities include continuous and dynamic fader control as well as fader start-type commands, which are essential for latency-critical WAN connections and to avoid speech truncation on air. The GUI’s feature set also includes sophisticated test mode and lineup patterns, as well as controllable DIM levels to the talents’ ear-pieces for talkback from the director. The graphical interface also allows monitoring the control connection to the host console, the on-air status, and Sync. You can also monitor Listen and PFL, as well as the local talkback system.
HC Bridge SRC
I/O & Interfaces
I/O & Interfaces
INTRAPLEX® ASCENT
The scalability of Ascent makes it ideal for applications that require multiple channels of audio encoding and decoding at head-end sites. The high-density solution reduces cost and provides a path for convergence of IT and broadcast infrastructure. The interoperability between Ascent and IP Link codecs provides a complementary solution for remote contribution and distribution use cases.
Product Details
The Ascent platform continues the Intraplex tradition of providing an unprecedented level of reliability for audio over IP application. Built with enhanced network reliability and security in mind, Ascent supports the Dynamic Stream Splicing technology of IP Link codec, which provides “hitless” protection for packet losses using the combination of time and network diversity for packet transmission, and FEC.
To further enhance reliability and security, the platform supports the SRT protocol encapsulation. SRT (Secure Reliable Transport) is an open-source protocol that provides low-latency, reliable and secure streaming of audio and video data. The reliability in SRT is accomplished by a built-in packet re-transmission scheme; the security of streams is handled by the built-in AES-128/256 payload encryption.
In addition to “hitless” protection, like the IP Link codecs, the Ascent also supports automatic failover between different incoming streams or stored audio.
This combination of capabilities gives Intraplex codecs a market-leading position in STL over IP technology and unmatched network transport reliability.
Ascent supports both physical audio interfaces (AES3, Analog) and AES67 (AoIP), with a maximum of 16 full-duplex or a total of 32 channels (combination of encode and decode). Physical audio interfaces can be ordered as 4- or 8-channel cards, supporting both AES3 and analog audio signals. The Ascent product is built on the Commercial-Off-The-Shelf (COTS) x86 architecture to leverage the scalability and cost of the technology. The product is available in a 1RU branded hardware server and as a software-only option.
Main Features of Intraplex® Ascent
- 4, 8, 16 stereo channels. Each channel can be full-duplex, encode-only or decode-only
- Supports AES3, Analog and AES67 audio input and outputs
- Standard Coding: Linear, Opus
- Optional: AAC-HE, AAC-HEv2, AAC- ELD, AAC-xHE, MPEG 2 and MPEG 3 audio coding
- Protocol Encapsulation: Real-time Transport Protocol (RTP), Secure Reliable Transport (SRT)
- Streams: Multicast, unicast, and multi-unicast
- 10 streams per channel with up to 20 destinations per transmit stream; maximum of 100 streams per system
- Three independent IP interfaces for redundant network operation
- Built-in silence sensor with optional stream switchover
- Automatic backup to audio playout from USB drive or external audio source
- Multicoding: can encode the same audio source in multiple formats
- Prioritized stream sources at the output with automatic switch over and switch back between primary and secondary streams and backup sources (streams, USB, external audio source)
- Programmable RTP level Forward Error Correction (FEC) scheme
- Programmable time diversity and Interleaving of streams to combat burst packet losses
- Integrated scheduler for automated scheduled program switching
- Integrated with Intraplex LiveLook (network analytics and monitoring software)
- Web and SNMP for management
- Multiple web account types to restrict access
- GPIOs: up to 32 in, 8 out. Available for stream transport and alarm assignment
- Network reliability
- Dynamic Stream Splicing with network and time diversity for “hitless” packet loss protection
- Programmable RTP level Forward Error Correction (FEC)
- Secure Reliable Transport (SRT): new protocol provides automatic packet retransmission method
- IP Security
- Access control per interface
- Encryption of streams with AES-128
IQOYA X/LINK-LE
IQOYA X/LINK-LE
Streamlined audio codec
IQOYA X/LINK-LE is a 1U rack streamlined IP audio codec designed for the delivery of a stereo source (or two mono sources) over IP networks for STL and SSL, but also DVB audio, or WEB radio. IQOYA X/LINK-LE benefits from all the major features of X/LINK but at an attractive price. It can be used in legacy analog or AES/EBU audio environments, as well as in full-IP audio infrastructures (AES67, Ravenna, Livewire), making it a good investment for the migration to IP audio. Like all the IQOYA products, X/LINK-LE is based on Fluid IP technology which offers the redundant dual streaming feature, allowing for reliable connections over inexpensive IP links. Based on a low consumption and fanless powerful hardware platform, IQOYA X/LINK-LE is designed for 24/7/365 use.
key points
Cost effective solution with essential features, and no compromise on reliability
Designed for audio service continuity and failsafe operation
Ongoing product support with flexible options
Invest now in a codec adapted to your current legacy audio infrastructure, and to your future full-IP audio infrastructure
Easy integration into existing SNMP based supervisors (SET, GET, Traps)
IQOYA X/LINK-DUAL
IQOYA X/LINK-DUAL
2 stereo codecs in 1U
IQOYA X/LINK-DUAL is a 1U rack IP audio codec designed for the delivery of two stereo sources (or four mono sources) over IP networks for STL and SSL, DVB audio, WEB radio, intercom and commentary. It can be used in legacy analog and AES/EBU audio infrastructures, as well as in full-IP audio infrastructures (AES67, Ravenna, Livewire), making it a perfect investment for the migration to in-house IP audio. Like all the IQOYA products, X/LINK-DUAL is based on Fluid IP technology which offers the redundant dual streaming feature, allowing for reliable connections over inexpensive IP links. Based on a low consumption, fanless and powerful hardware platform, IQOYA X/LINK-DUAL is designed for 24/7/365 use.
Key points
Space efficient: two stereo codecs in a 1U rack, with simultaneous delivery to transmitter sites, WEB radio CDNs, DVB multiplexers, and other studios
Adapted to your current legacy audio infrastructure, and to your future full-IP audio infrastructure
Easy integration into existing SNMP based supervisors (SET, GET, Traps)
Ongoing product support with flexible options
