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USER’S MANUAL
Section 6: AUDIO CODING REFERENCE 115
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.
Summary of Contents for Zephyr Xstream
Page 2: ......
Page 26: ...USER S MANUAL Section 1 QUICK RESULTS 14...
Page 30: ...USER S MANUAL Section 2 INTRODUCTION Getting to Know the Zephyr Xstream 18...
Page 70: ...USER S MANUAL Section 3 GUIDED TOUR of the HARDWARE 58...
Page 144: ...USER S MANUAL Section 6 AUDIO CODING REFERENCE 132...
Page 164: ...USER S MANUAL Section 8 LIVEWIRETM IP Audio 152...
Page 310: ...USER S MANUAL Appendix 1 Codec Interoperability Information 298...
Page 320: ...USER S MANUAL Appendix 3 ISDN Cause Phrases Values 308...
Page 324: ...USER S MANUAL Appendix 4 Known Working SPID Formats by Telco 312...
Page 356: ...USER S MANUAL Appendix 9 Modular Cable Guide 344...