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 Zephyr Iport
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
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.
Energy-Onix Tele-link III
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