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A break from Tradition: Using IP for an STL
Harris/Intraplex is well-known in the STL field. A fairly new product is called Netxpress, which basically uses the same form-factor as the legacy T1/E1 shelf. This unit accepts the current line of Intraplex audio, video and data modules, and connects to the far end via IP. It has a total payload bandwidth of 8Mb/s and can accommodate up to 32 data streams (point-to-point unidirectional or bidirectional; or point-to-multipoint unidirectional multicast). It comes with a network management system that allows the user to control the packet size on a per-stream basis; a packet-jitter buffer that allows the user (also on a per-stream basis) to minimize the negative effects of issues such as packet delay, and the restoration of out-of-sequence packets; priority tagging, which can be used to give audio stream payload packets a higher-priority; and finally user-adjustable forward error correction (FEC) that is used to help the far end rebuild lost or dropped audio packets. Network statistics are available for current and cumulative packets sent, received, lost and delayed on a per-stream basis.
Musicam offers the Suprima-a 1RU, two-channel analog or AES interface audio codec (full-duplex). Communication with the far end is accomplished via ISDN or IP. Compression algorithms supported include MPEG 1/2 (layers 2 and 3), AAC LC, AAC LD, Apt-x (standard and enhanced), uncompressed PCM, G.722 and G.711 (with echo cancellation). The two channels can operate completely independent of one another if the compression algorithm is G.711, G.722 or MPEG. The send and receive directions can use different algorithms. Auxiliary contact closures are available in all MPEG modes. IP protocols supported include TCP, UDP and real time audio streaming. The Suprima has a built-in Web server so a browser can control it remotely. I don't want to forget the front-panel headphone jack that can be used to monitor audio in either direction. That's always a handy feature.
Telos is making its presence known in this field and has recently introduced the Zephyr Iport. This 2RU device can send eight stereo audio feeds over IP networks. It uses the Livewire standard for networked audio over Ethernet and typically would be part of an Axia IP audio network. (If the unit isn't part of an Axia IP-audio network, use of the Iport will require the acquisition of an Axia AES or analog audio node.) Compression algorithms include AAC LD, AAC and MPEG 3 (layers 2 and 3). Full configuration, and remote control is done via the embedded Web browser.
Perhaps you want to ease in to the whole audio-over-IP technology; if so then the product line from Barix may be exactly what you are looking for. The Instreamer 100 is a small, stand-alone audio encoder that connects to the far end via IP. It makes use of the MPEG 3 compression algorithm (16 to 48kHz sample rate, up to 192kb/s variable bit-rate) with stereo audio, RCA inputs or coaxial or optical S/PDIF. Control is accomplished via embedded web browser or RS-232. The complementary decoder is the Exstreamer 100: This unit will decode MPEG 3 (up to 320kb/s fixed or variable bit-rate) or Windows Media encoder (up to 384kb/s). Audio outputs are delivered via RCA connectors; control is done via embedded Web browser or RS-232.
There are several other players to consider — some that you may have not previously thought of. The first is a company well known for making transmitters: Energy Onix. Its offering in this field is the Tele-link III. This is a single-rack unit codec built on top of a small industrial computer running Linux. Audio inputs and outputs are balanced analog; the network connection is handled through an RJ-45, connecting at 10 or 100Base T. The necessary data rate is 128kb/s for 48kHz sampling, with a 16-bit word, by way of either the MP3 or Ogg-Vorbis compression algorithms. All control is done by way of the front panel.
MDO-UK also has a single rack unit solution for audio over IP. Its product is known as Audio-TX STL-IP. Audio inputs and outputs are done by way of balanced AES. The unit will accept wordclock. The codec can generate up to six streams, while receiving audio from one remote location (TCP/IP, UDP or UDP multicast). Audio can be sent in an uncompressed fashion (assuming bandwidth can accommodate it) or at reduced data rates making use of any one of the following compression algorithms: MPEG 2 layers 2 or 3, ADPCM, AAC, AAC-LD or AACPlus. FEC is built-in. The configuration and control are done via a Web browser.
Audio over IP for STL applications is not a new idea, by any means. Having LAN and/or WAN connectivity at a transmitter site, even one that is way out in the sticks, is becoming more and more common — and we've gotten to the point where we expect just about every electronic device to have some sort of network connection. Even though the world is going this way, I'm not ready to hand over my main STL to a contentious network just yet. Still, as time goes by, it's conceivable that type of network will provide the same level of reliability, all things considered, as the type of networks and links we use today. Now might be a good time for you to learn how it's done.
Irwin is the chief engineer of WKTU-FM, New York City.
+44 121 256 0200
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