AUDIO CODING REFERENCE | 69
MPEG
By far, the most popular high fidelity audio coders rely upon techniques developed under the MPEG umbrella.
MPEG stands for Motion Pictures Expert Group, a joint committee of the International Standards Organization
(ISO) and the International Electrotechnical Commission. Three decades ago, when the CD had just been
introduced, the first proposals for audio coding were greeted with suspicion and disbelief. There was widespread
agreement that it would simply not be possible to satisfy golden ears while deleting 80% or more of the digital audio
data. But the audio coding pioneers were persistent, and the MPEG audio group was formed. Since 1988, they have
been working on the standardization of high quality audio coding. Today, almost all agree not only that audio bit
rate reduction is effective and useful, but that the MPEG process has been successful at picking the best technology
and encouraging compatibility across a wide variety of equipment.
The MPEG process is open and competitive. A committee of industry representatives and researchers meet to
determine goals for the target bit rate, quality levels, application areas, testing procedures, etc. Interested organiza-
tions that have something to contribute are invited to submit their best work. A careful double blind listening test
series is then conducted to determine which of the entrant’s technologies delivers the highest performance. The
subjective listening evaluations are done at various volunteer organizations around the world that have access to
both experienced and inexperienced test subjects. Broadcasters are the most common participants, with many of
the important test series conducted at the BBC in England, the CBC in Canada, and NHK in Japan. Finally, results
are tabulated, a report is drafted, and ultimately a standard is issued.
In 1992, under MPEG1 (the first of the MPEG standards), this process resulted in the selection of three related
audio coding methods, each targeted to different bit rates and applications. These are the famous layers: 1, 2, and 3.
As the layer number goes up, so does performance and implementation complexity. Layer 1 is not much used. Layer
2 is widely used for Digital Audio Broadcasting (DAB) in Europe, audio for video, and broadcast play-out systems.
Layer 3 – which Telos was the first to use in the Zephyr – is widely used in broadcast codecs and has gone on to sig-
nificant Internet and consumer electronics fame under the moniker derived from the file extension: MP3. MPEG2
opened the door for new work, and some minor improvements were added to both Layers 2 and 3. In 1997, the first
in the AAC family was added to the MPEG2 standard. MPEG4 is ongoing now, but it has already been decided that
AAC will be the “general audio” coder under this umbrella. (MPEG3 was skipped for reasons unknown)
MPEG4 AAC (Advanced Audio Coding)
The MPEG4 AAC system is the newest audio coding method selected by MPEG. It is a fully state-of-the-art audio
compression tool that provides performance superior to any known approach at bitrates greater than 64kbps, and
excellent performance relative to the alternatives at bitrates reaching as low as 16kbps.
The idea that lead to AAC was not only to start fresh, but also to combine the best work from the world’s leading
audio coding laboratories: Fraunhofer, Dolby, Sony, and AT&T were the primary collaborators that offered com-
ponents for AAC. The hoped-for result was ITU (International Telecommunications Union) “indistinguishable
quality” at 64 kbps per mono channel. That is, quality indistinguishable from the original, with no audio test item
falling below the “perceptible, but not annoying” threshold in controlled listening tests.
The MPEG test items include the most difficult audio known to codec researchers, so this was a daunting challenge.
The thinking was that if a codec could pass this test, it would surely be transparent for normal program material
like voice and pop music, which are much easier to encode. AAC designers chose to use a new modular approach
for the project, with components being plugged in to a general framework, in order to match specific application
requirements and the always-present performance/complexity/delay tradeoffs.
Compared to the previous layers, AAC takes advantage of such new tools as temporal noise shaping, back-
ward-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.
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