Telos Zephyr Xstream User Manual page 127

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
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, 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.
AAC is a lot more sophisticated than the previous MPEG layers 2 &3, providing significantly more
coding power. Because it uses a modular approach, new ideas can be developed and plugged in
to the basic structure. This had the additional advantage that it was possible to combine the
various components from different developers, taking the best pieces from each. AAC was built
on a similar structure to Layer‐3, and thus retains most of its features. Nevertheless, compared
to the previous MPEG layers, AAC benefits from some important new additions to the coding
toolkit:
•
An improved filter bank with a frequency resolution of 2048 spectral
components, nearly four times the number used by Layer‐3.
•
Temporal Noise Shaping, a new and powerful element that minimizes the
effect of temporal spread. This benefits voice signals, in particular.
•
A Prediction Module guides the quantizer to very effective coding when
there is a noticeable signal pattern, like high tonality.
•
Perceptual Noise Shaping allows a finer control of quantization resolution,
so bits can be used more efficiently.
Because it uses this modular approach, an implementer may pick and choose among the
component tools to make a product with appropriate performance/complexity ratios. Or, new
modules can be developed later and "plugged in" to its basic structure. Three default profiles
have been defined, using different combinations of the available tools:
•
Main Profile. Uses all tools except the gain control module. Provides the
highest quality for applications where the amount of random accessory
memory (RAM) needed is not constrained.
•
Low‐complexity Profile. Deletes the prediction tool and reduces the
temporal noise‐shaping tool in complexity.
•
Sample‐rate Scaleable (SRS) Profile. Adds the gain control tool to the low
complexity profile. Allows the least complex decoder.
USER'S MANUAL
Section 6: AUDIO CODING REFERENCE 115