User manual

• 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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