Sound field programs, M cinema dsp, Silent cinema – Yamaha HTR-5860 User Manual

Page 96: Virtual cinema dsp, Audio information, M itu-r, Lfe 0.1 channel, Pcm (linear pcm), Sampling frequency and number of quantized bits, Sound field programs audio information

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GLOSSARY

Sound field programs

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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

mea.sured data, YAMAHA CINEMA DSP uses
YAMAHA original sound field technology to combine
Dolby Pro Logie, 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, reali,stic 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 enjoyed on headphones.

■ 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.

Audio information

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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 frequency 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
technique for sampling the size of the analog signal per

very small unit of time. Standing for ’'pufse 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 w hen

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 frequency, 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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