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•
CINEMA
DSP
Since the Dolby Surround and DTS systems were
originally designed for use in movie theaters, their effect
is best felt in a theater having many speakers and designed
for acoustic effects. Since home conditions, such as room
size, wall material, number of speakers, and so on, can
differ so widely, it's inevitable that there are differences in
the sound heard as well. Based on a wealth of actually
measured data, YAMAHA CINEMA DSP uses
YAMAHA original sound field technology to combine
Dolby Pro Logic, Dolby Digital and DTS systems to
provide the visual and audio experience of movie theater
in the listening room of your own home.
•
SILENT CINEMA
YAMAHA
has developed
a natural,
realistic
sound effect
DSP algorithm
for headphones.
Parameters
for headphones
have been set for each sound
field so that accurate
representations
of all the sound field
programs
can be enioyed
on hea@hones.
•
Virtual CINEMA DSP
YAMAHA
has developed
a Virtual CINEMA
DSP
algorithm
that allows
you to enjoy DSP sound field
surround
effects even without
any surround
speakers
by
using virtual
surround
speakers.
It is even possible
to enjoy Virtual CINEMA
DSP using a
minimal
two-speaker
system
that does not include
a center
speaker.
•
ITU-R
ITU-R is the radio communication sector of the ITU
(International
Telecommunication
Union). ITU-R
recommends a standard speaker placement which is used
in many critical listening rooms, especially for mastering
purposes.
•
LFE
0.1 channel
This channel
is for the reproduction
of low bass signals.
The fi'equency
range
for this channel
is 20 Hz to 120 Hz.
This channel
is counted
as 0.1 because
it only enforces
a
low frequency
range compared
to the full-range
reproduced
by the other 5/6 channels
in Dolby Digital
or
DTS 5.1/6.1-channel
systems.
•
PCM (Linear PCM)
Linear
PCM is a signal format
under which
an analog
audio
signal is digitized,
recorded
and transmitted
without
using any compression.
This is used as a method
of
recording
CDs and DVD audio. The PCM system
uses a
techniqne
lbr smnpling
the size of the analog signal per
very small unit of time. Standing
for "pulse
code
modulation",
the analog signal is encoded
as pulses
and
then modulated
for recording.
•
Sampling frequency
and
number
of
quantized
bits
When digitizing
an analog audio signal,
the number
of
times the signal is sampled
per second
is called the
sampling
frequency,
while the degree of fineness
when
converting
the sound level into a numeric
value is called
the number
of quantized
bits.
The range
of rates that can be played back is determined
based on the sampling
rate, while
the dynamic
range
representing
the sound level difference
is determined
by
the number
of quantized
bits. In principle,
the higher the
sampling
fi'equency,
the wider the range of frequencies
that can be played back, and the higher
the number
of
quantized
bits, the more finely the sound level can be
reproduced.
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