Audio Information (Others) - Yamaha RX-V781 Owner's Manual

DTS Neo:6
DTS Neo:6 enables 6-channel playback from 2-channel sources. There are two modes available: "Music mode" for
music sources and "Cinema mode" for movie sources. This technology provides discrete full-bandwidth matrix
channels of surround sound.
DTS:X
DTS:X is the next generation object-based, multi-dimensional audio technology from DTS. Unbound from channels,
DTS:X conveys the fluid movement of sound to create an incredibly rich, realistic and immersive soundscape - in
front of, behind, beside and above the audience - more accurately than ever before. DTS:X offers the ability to
automatically adapt the audio to the speaker layout that best fits the space, from a television's built-in speakers to
a home surround theater system to a dozen or more speakers in a commercial cinema. Immerse yourself at
www.dts.com/dtsx
FLAC
FLAC is a file format for lossless audio data compression. FLAC is inferior to lossy compressed audio formats in
compression rate but provides higher audio quality.
MP3
One of the compressed digital audio format used by MPEG. With psychoacoustic technologies, this compression
method achieves a high compression rate. Reportedly, it is capable of compressing data quantity by about 1/10
maintaining a certain level of audio quality.
MPEG-4 AAC
An MPEG-4 audio standard. It is used for mobile telephones, portable audio players, and audio streaming services on
Internet because it allows a high compression rate of data while maintaining better audio quality than MP3.
Neural:X
Neural:X is the latest downmixing/upmixing and spatial remapping technology from DTS. It is built in to DTS:X to
provide upmix of Neural:X-encoded and non-encoded (PCM) data. In DTS:X for AVRs and Sound Bars, Neural:X can
produce up to 11.x channels.
PCM (Pulse Code Modulation)
PCM is a signal format under which an analog audio signal is digitized, recorded, and transmitted. This technology is
the basis of all other audio format. This technology is used as a lossless audio format called Linear PCM for audio on
a variety of media, including CDs and BD (Blu-ray discs).
Sampling frequency/Quantization bit
Sampling frequency and quantization bits indicate the quantity of information when an analog audio signal is
digitized. These values are noted as in the following example: "48 kHz/24-bit".
• Sampling frequency
Sampling frequency (the number of times the signal is sampled per second) is called the sampling rate. When the
sampling frequency is higher, the range of frequencies that can be played back are wider.
• Quantization bit
The number of quantization bits indicate the degree of accuracy when converting the sound level into a numeric
value. When the number of quantized bits is higher, the expression of the sound level is more accurate.
WAV
Windows standard audio file format, which defines the method of recording the digital data obtained by converting
audio signals. By default, the PCM method (no compression) is used, but you can also use other compression
methods.
WMA (Windows Media Audio)
One of the compressed digital audio formats developed by Microsoft Corporation. With psychoacoustic
technologies, this compression method achieves a high compression rate. Reportedly, it is capable of compressing
data quantity by about 1/20 maintaining a certain level of audio quality.

Audio information (others)

Bi-amplification connection (Bi-amp)
A bi-amplification connection uses two amplifiers for a speaker. When you use the bi-amplification connection, the
unit drives the tweeter and woofer in a speaker with the discrete amplifiers. As a consequence, the tweeter and
woofer provide clear audio signal without the interference.
LFE (Low Frequency Effects) 0.1 channel
This channel reproduces low-frequency bass signals and has a frequency range from 20 Hz to 120 Hz. This channel is
added to the channels for all bands with Dolby Digital or DTS to enhance low frequency audio effects. This channel
is labeled 0.1 because it is limited to only low frequency audio.
Lip sync
Video output sometimes lags behind audio output due to the complexity of signal processing caused by an increase
in video signal capacity. Lip sync is a technique for automatically correcting the timing lag between audio and video
output.
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