An introduction to MPEG AAC (Advanced Audio Coding)
The Telos Zephyr Xstream takes advantage of the newest, most efficient coding method yet devised, MPEG2 AAC (Advanced Audio Coding). Its benefits to users are numerous, and include the ability to output compressed audio indistinguishable from source at 128kbps, and low-delay modes which provide fidelity comparable to MPEG Layer III at up to 30 percent faster on-the-fly encoding rates.
AAC Technical Description
AAC is the newest audio coding method selected by MPEG and became an international standard in April 1997. It is a fully state-of-the-art audio compression tool kit that provides performance superior to any known approach at bit rates greater than 64 kbps and excellent performance relative to the alternatives at bit rates reaching as low as 16 kbps.
The development of AAC began when researchers became convinced that significant improvements would be possible by abandoning backward compatibility to the earlier MPEG layers. The idea was to start fresh and take the best work from the world’s leading audio coding laboratories. Fraunhofer Institute, Dolby, Sony and AT&T were the primary collaborators. The hoped for result was International Telecommunications Union (ITU)-R indistinguishable quality at 64 kbps per mono channel. This was a fairly daunting requirement because it requires that no test item fall below the perceptible, but not annoying threshold in controlled listening tests.
The test items include the most difficult-to-encode audio known to researchers—isolated pitch pipe, harpsichord and glockenspiel recordings, among others. The thinking was that if a coding system passes this requirement, it will almost certainly perform well with normal program material. Pop or western classical music is tremendously easier to encode.
Compared to the previous layers, AAC takes advantage of such new tools as temporal noise shaping, backward adaptive linear prediction and enhanced joint stereo coding techniques. AAC supports a wide range of sampling rates (8–96 kHz), bit rates (16–576 kbps) and from one to 48 audio channels.
The AAC system uses a modular approach. An implementer may pick and choose among the component tools to produce a system with appropriate performance-to-complexity ratios.
AAC is the first codec system to fulfill the ITU-R/EBU requirements for indistinguishable quality at 128 kbps/stereo. It has approximately 100% more coding power than Layer II and 30% more power than the former MPEG performance leader, Layer III.
AAC takes advantage of such tools as temporal noise shaping, backward adaptive linear prediction and enhanced joint stereo coding techniques in addition to the techniques used in ISO/MPEG Layer III. Zephyr Xstream is the first broadcast codec to incorporate the power of AAC coding, resulting in superior high-fidelity audio at lower bitrates and with less delay than Layer III or Layer II.
AAC is the most powerful coding method available in Zephyr Xstream, and we particularly like it because it is perfectly matched to the bitrates available on ISDN BRI lines.
15 or 20kHz mono or stereo audio bandwidth.
Full-fidelity mono on a single 56/64kbps channel.
Near-CD quality stereo on a single ISDN Telco circuit.
Affordable, transparent, audio transmission for AM/FM radio or television audio.
LD-AAC (Low Delay AAC)
Zephyr users have known for years that Layer III offers all the fidelity needed in most broadcast situations. However, they also know that the delay of layer III can be frustrating, particularly if high fidelity is needed in both directions.
The folks at Fraunhofer were aware of these factors, and have developed an extension to AAC called "Low Delay AAC" or "LD-AAC" for short. LD-ACC offers quality equivalent to Layer III with about 75% less delay! Zephyr Xstream is the first codec to implement LD-AAC, and we think you will find it very useful.