USER’S MANUAL
Section 6: AUDIO CODING REFERENCE 118
AAC-LD (AAC Low Delay)
When
announcers
use
codecs
for
broadcast
remote
applications,
they
often
need
to
have
natural
two
‐
way
interaction
with
other
program
participants
located
back
at
the
studio,
or
callers.
Because
it
is
a
hot
topic
for
engineers
working
in
the
field
of
Internet
telephony,
a
number
of
studies
have
been
conducted
to
determine
user
’s
reactions
to
delays
in
telephone
conversations.
The
data
apply
directly
to
the
application
of
professional
codecs
to
remotes,
so
it
is
interesting
to
take
a
peak
over
the
shoulder
of
the
telecom
guys
to
see
what
they
have
learned.
For
broadcast
remotes,
we
try
to
arrange
our
system
so
that
there
is
no
path
for
the
field
announcer’s
voice
to
return
to
his/her
headphones.
Nevertheless,
sometimes
echo
is
unavoidable.
For
example,
this
can
happen
when
a
telephone
hybrid
has
leakage
or
when
a
studio
announcer
has
open
‐
air
headphones
turned
‐
up
loud
and
the
audio
makes
its
way
into
the
studio
microphone.
When
there
is
no
echo,
it
has
been
discovered
that
anything
less
than
100
ms
one
‐
way
delay
permits
normal
interaction
between
participants.
Between
100
and
250
ms
is
considered
“acceptable.”
ITU
‐
T
standard
G.114
recommends
150
ms
as
the
maximum
for
“good”
interactivity.
Echo
introduces
a
different
case.
As
you
might
expect,
echo
is
more
or
less
annoying
depending
upon
both
the
length
of
time
it
is
delayed
and
its
level.
Telephone
researchers
have
measured
and
quantified
reactions,
and
ITU
‐
T
G.131
reports
the
findings
and
makes
recommendations.
Summary of ITU-T G.13, with recommendations for designers of telephone systems that must cope
with echo. This shows Talker Echo Loudness Rating vs delay.
There
are
codecs
using
other
than
perceptual
technologies
that
have
lower
delay,
but
they
are
not
as
powerful.
That
is,
for
a
given
bitrate,
they
do
not
achieve
fidelity
as
good
as
the
MPEG
systems
we
have
been
examining.
The
common
G.722
is
an
example.
It
uses
ADPCM
(Adaptive
Delta
Pulse
Code
Modulation),
which
can
have
delay
as
low
as
10
ms,
but
with
much
poorer
quality.
So
the
question
arises:
Is
it
possible
to
have
high
quality
and
low
delay
in
the
same
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...
