Section 7 AUDIO CODING PRINCIPLES
106
OVERVIEW
Introduction to Audio Coding
Without data rate reduction, high quality audio requires a transmission capacity of
about 700kbps for each audio channel. At 56kbps, a telephone channel offers a rate
about 8% of the Compact Disc’s.
The first practical coding methods used a principle called ADPCM, Adaptive Delta Pulse
Code Modulation. This takes advantage of the fact that it takes fewer bits to code the
difference, or delta, between successive audio samples compared to using the
individual values. Further efficiency is had by adaptively varying the difference
comparitor according to the nature of the program material. G.722 and APT- X are
examples of ADPCM schemes. They achieve around a factor of 4 reduction in bitrate.
For high- fidelity transmission, algorithms with more power are required. These are
based on psychoacoustics, where the coding process is adapted to the human
perceptual system. There are several algorithms available with varying complexity and
performance levels.
Some years ago, the international standards group ISO/IEC established the ISO/MPEG
(Moving Pictures Expert Group)
in order to develop a universal standard for encoding
moving pictures and associated audio for use with digital storage and transmission
media. The standard was finalized in November 1992 with three related algorithms,
called Layers, being defined for encoding of audio taking advantage of psychoacoustic
effects. While Layer I and II are intended for compression factors of about 4 and 6- 8
respectively, with Layer III factors of up to 12.5 can be achieved.
Basic Principles of Perceptual Coding
With perceptual coding, only information that can be perceived by the human auditory
system is retained.
Lossless – which, for audio, translates to noiseless – coding with perfect reconstruction
would be an optimum system, since no information would be lost or altered. It might
seem that lossless, redundancy- reducing methods (such as PKZIP, Disk Doubler, and
others used for computer hard- disk compression) would be applicable to audio.
Unfortunately, no constant compression rate is possible due to signal- dependent
variations in redundancy: There are highly redundant signals like constant sine tones
(where the only information necessary is the frequency, phase, amplitude, and duration
of the tone), while other signals, such as those which approach broadband noise, may
be completely unpredictable and contain no redundancy at all. Since any system
intended for a telephone channel is required to have a consistent output rate and must
accommodate the worst case, no compression is possible with redundancy reduction
alone.
Fortunately, psychoacoustics permits a clever solution! Effects called “masking” have
been discovered in the human auditory system. These masking effects (which merely
prove that our brain is also doing the equivalent of coding) have been found to occur in
both the frequency and time domains and can be exploited for audio data reduction.
Summary of Contents for Zephyr
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Page 39: ...39 SECTION 3 ZEPHYR AT A GLANCE ...
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Page 53: ...Section 4 INSTALLATION BASIC OP 53 SECTION 4 INSTALLATION BASIC OPERATION ...
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Page 85: ...Section 5 ISDN 85 SECTION 5 ISDN ...
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Page 119: ...Section 8 DETAILED MENU REFERENCE 119 SECTION 8 DETAILED MENU REFERENCE ...
Page 157: ...Section 9 REMOTE CONTROL 157 SECTION 9 REMOTE CONTROL ...
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Page 177: ...Section 10 ADVANCED PROBLEM SOLVING 177 SECTION 10 ADVANCED PROBLEM SOLVING ...
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Page 197: ...Section 11 TECHNICAL INFORMATION 197 SECTION 11 DETAILED TECHNICAL INFORMATION ...
Page 219: ...Section 12 SCHEMATICS 219 SECTION 12 SCHEMATICS ...
Page 221: ...Section 13 MANUFACTURER S DATA SHEETS 221 SECTION 13 MANUFACTURER S DATA SHEETS ...
Page 223: ...Section 14 SPECIFICATIONS WARRANTY 223 SECTION 14 SPECIFICATIONS AND WARRANTY ...
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