A
u d i o T X S T L - I P
F e a t u r e s & S p e c i f i
c a t i o n s . . . |
AudioTX
STL-IP Classic IP codec provides for live audio transmission over
IP networks with transmission grade audio quality & robustness
and extremely low delays - as low as 5ms!
- IP
Codec for live audio: Transmit
and receive audio using point-to-point UDP or TCP/IP, and
point-to-multipoint UDP Multicast network protocols over
ANY IP Network - including private networks (LAN/WAN), Satellite,
Wireless networks, T1/E1, ATM or the Internet.
- A
single STL-IP Classic system can transmit audio on up to six simultaneous
connections, each using different audio coding and network
protocols if required. Using Multicast, audio can be sent
to an unlimited number of destination units. Audio can be
received from one remote location, independently from transmission.
- AudioTX
STL –IP works with linear (uncompressed) audio at
up to 24 bits and 96kHz sample rate, or compressed audio
via built-in Professional Grade MPEG Layer 2 or MPEG Layer
3 coding/decoding, J.41, DAT12, ADPCM, G.722 or our extra Low-Bitrate
speech codec. Plus MPEG4 AAC, AAC Low-Delay
and HE-AAC v2 with the optional AAC Coding Pack for Stereo
audio from just 14kbps!
- Optional
APTx Coding (Enhanced APTx, 16 and 24 bits) with the APTx
Codec Pack.
- Selectable
Forward Error Correction (FEC) and network jitter compensation
where required.
- Synchronous
transmission of serial ancillary data and/or contact closures
(TTL GPIO).
- Built-in
silence and audio overload detectors.
- Monitor
and control via web-browser control interface, SNMP traps
and queries, E-mail alerts, Telnet style IP remote control
interface (using simple text commands and responses), included
software and logic level status outputs.
- Incredibly
flexible and cost-effective solution.
AudioTX
STL-IP can send live audio using up to 6 simultaneous connections
- including any combination of UDP, TCP/IP or UDP Multicast
(to an unlimited number of destinations for each UDP Multicast
connection). The IP Codec system can receive audio from one
location.
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AUDIO
SPECIFICATIONS AND PROTOCOLS |
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Summary:
Professional
grade analogue balanced Stereo audio inputs and outputs plus
AES/EBU digital audio in/out, external wordclock input. Audio
in/out at up to 24 bit, 96 kHz sample rate |
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Mono/Stereo
audio transmit/receive using Linear (uncompressed) audio,
Broadcast Quality MPEG Layer 2, MPEG Layer 3, J.41, Mono,
Stereo, Joint-Stereo, Dual-Mono operation.
MPEG4 AAC, AAC Low-Delay and HE-AAC v2 with the optional AAC
Coding Pack.
Enhanced APTx coding with the APTx Codec Pack.
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Detailed
specification: |
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Linear
(uncompressed) audio: |
Uncompressed audio at 8kHz to 96kHz sample rate, 16 or 24 bit.
Mono or Stereo modes. |
Full-bandwidth
linear audio with a 5ms delay. |
| MPEG4
AAC: |
Professional
grade AAC coded audio at between 16 and 48 kHz sample rate,
16 bit, Mono, Stereo, Joint-Stereo and dual-mono modes. 24-320kbps. |
Provides
full-bandwidth broadcast quality stereo audio at bitrates between
64kbps and 320kbps. Near transparent Stereo audio from 64kbps.
150ms delay. |
| MPEG4
AAC-Low Delay: |
Professional
grade Low Delay AAC at 16 to 48 kHz sample rate, 16 bit, Mono,
Stereo, Joint-Stereo and dual-mono modes. Low Delay version.
24-320kbps. |
Provides
full-bandwidth broadcast quality stereo audio at bitrates between
80kbps and 320kbps with just 40ms delay. |
MPEG4
HE-AAC and HE-AAC v2:
(High
Efficiency AAC, AACPlus): |
Offering
expectional audio quality at very low bitrates. HE-AAC coded
audio at between 32 and 48 kHz sample rate, 16 bit, Mono, Stereo,
and Parametric Stereo. 14 to 96 kbps. |
Provides
full-bandwidth excellent quality stereo audio at bitrates between
14kbps and 96kbps. 260ms delay. |
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| MPEG
Layer 2 coded audio: |
Professional
MPEG Layer 2 coded audio at between 16 and 48 kHz sample rate,
16 bit, Mono, Stereo, Joint-Stereo and dual-mono modes. |
Provides
full-bandwidth broadcast quality stereo audio at bitrates between
128kbps and 384kbps. Mono audio from 64kbps. 45ms delay. |
| MPEG
Layer 3 coded audio: |
Professional
MPEG Layer 3 coded audio at between 16 and 48 kHz sample rate,
16 bit, Mono, Stereo, Joint-Stereo and dual-mono modes. |
Provides
full-bandwidth broadcast quality stereo audio at bitrates between
128kbps and 384kbps. Mono audio from 64kbps. 125ms delay. |
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| J.41: |
High-grade
audio compression for near transparent professional audio, 32kHz
sample rate, Mono/Stereo. |
Mono
audio at 384kbps, Stereo at 768kbps. 5ms delay. |
| ADPCM: |
Professional
quality compression, 32kHz or 48kHz sample rate, Mono or Stereo
mode. |
Mono
audio at 128kbps or 192kbps, Stereo at 256kbps or 384kbps. 5ms
delay. |
| G.722: |
Good
quality algorithm for speech/voice with a 7.5kHz audio bandwidth.
Runs at 16kHz sample rate. |
Mono
audio at 64kbps. 5ms delay. |
| LB-1: |
Extra
Low-Bitrate speech codec offering 7.5kHz audio bandwidth. Runs
at 16kHz or 24kHz sample rate and a range of bitrates determine
quality. |
Mono
audio from 12kbps. 40ms delay. |
| APTx: |
Enhanced
APTx Coding - low delay, high quality compressed audio, choice
of 16 or 24 bits. |
Bitrates
range from 64kbps to 576kbps. 9ms delay. |
| Source
audio selection: |
User-selectable
channel inputs to audio transmission modules - Left channel,
Right Channel, Stereo or MonoMix (L+R).
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NETWORK CAPABILITIES AND SPECIFICATIONS |
MONITORING/CONTROL |
| Supports
all IP networks including Telco, MPLS, Private/Dedicated circuits,
LAN/WAN, Satellite, Wireless (incl. WiFi), ATM, T1/E1 and
The Internet for IP codec operation.
Network modes: UDP, TCP/IP, UDP Multicast modes
Audio transmit/receive bitrates between 24 kB/s and 4.6 mB/s
Optional transmission of ancillary serial data at up to 57600
bps, up to 4 in / 4 out GPIO (contact closures)
Optional use of FEC (forward error correction) and/or network
jitter compensation/safety buffer configurable in 1ms increments
from zero to 5 seconds
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Monitoring
and control via:
-
Web-browser control interface
- SNMP
traps and queries
- E-mail
alerts
- Telnet
style IP remote control interface (using simple text commands
and responses)
- Included
software
- Logic
level (TTL) status outputs
Built-in
silence and audio overload detectors. |
AUDIO,
NETWORK & DATA CONNECTIONS |
|
Analog
audio inputs: |
Stereo
balanced inputs, 2x XLR (F) |
-18db
nominal signal level. +18db at analog inputs = 0dbFS (digital
full scale). |
| Analog
audio outputs: |
Balanced
Stereo outputs, 2x XLR (M) |
-18db
nominal signal level. 0dbFS (digital full scale) = +18db at
analog inputs. |
| Digital
audio input: |
AES/EBU
digital input, XLR (F) |
Input
accepts both AES/EBU and SPDIF type of signals. |
| Digital
audio output: |
AES/EBU
digital output, XLR (M) |
|
| Clock
input: |
Wordclock
input, BNC. |
System
clock-source is user-selectable: internal clock, wordclock input
or use clock from incoming AES/EBU source. |
| GPIO: |
TTL
level inputs (4) and outputs (4) plus an additional 4 status
output signals, D-Sub 25 pin connector. |
GPIO
TTL inputs & outputs provide end-to-end transmission of
signals from transmitting to receiving units. |
| Ancillary
Data: |
RS-232
serial connection for ancillary data in and out, D-Sub 9 pin
connector. |
Serial
data can be transmitted/received alongside audio at up to 57600
bps. |
| Network
Connection: |
10/100/1000 Gigabit Ethernet (RJ45 connector) |
For TX, RX audio and web-management interface. |
AC
Power: |
96-264
VAC, 50-60Hz autosensing for worldwide operation.
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