Yamaha RX-N600 Benutzerhandbuch

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GLOSSARY
ADDITIONA
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INF
O
RMA
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IO
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LFE 0.1 channel
This channel reproduces low-frequency signals. The 
frequency range of this channel is from 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.
MP3
One of the audio compression methods used by MPEG. It 
employs the irreversible compression method, which 
achieves a high compression rate by thinning out the data 
of hardly audible part to the human ears. It is said to be 
capable of compressing the data quantity by about 1/11 
(128 kbps) while maintaining a similar audio quality to 
music CD.
Neo:6
Neo:6 decodes the conventional 2-channel sources for 6- 
channel playback by the specific decoder. It enables 
playback with the full-range channels with higher 
separation just like digital discrete signal playback. There 
are two modes available: “Music mode” for music sources 
and “Cinema mode” for movie sources.
Neural Surround
Neural Surround
 represents the latest advancement in 
surround technology and has been adopted by XM 
Satellite Radio for digital radio broadcast of surround 
recordings and live events in surround sound. Neural 
Surround
 employs psychoacoustic frequency domain 
processing which allows delivery of a more detailed sound 
stage with superior channel separation and localization of 
audio elements. System playback is scalable from 5.1 to 
7.1 multi-channel surround playback.
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 “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 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.
WAV
Windows standard audio file format, which defines the 
method of recording the digital data obtained by 
converting audio signals. It does not specify the 
compression (coding) method so a desired compression 
method can be used with it. By default, it is compatible 
with the PCM method (no compression) and some 
compression methods including the ADPCM method.
WMA
An audio compression method developed by Microsoft 
Corporation. It employs the irreversible compression 
method, which achieves a high compression rate by 
thinning out the data of hardly audible part to the human 
ears. It is said to be capable of compressing the data 
quantity by about 1/22 (64 kbps) while maintaining a 
similar audio quality to music CD.