Yamaha RX-V557 Owner's Manual

■

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.

■

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 enjoyed on headphones.

■

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 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.

■

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.

■

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 “pulse code 
modulation”, the analog signal is encoded as pulses and 
then modulated for recording.

■

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 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.