AUDIO COMPRESSION

ASHISH SHARMA
ROLL NO. E6802A21
REGD NO.-10801051

LOVELY PROFESSIONAL UNIVERSITY

Abstract: In this term paper I would like to present an
overview of AUDIO COMPRESSION. There is no
other thing that can compare to the versatility of audio
compression with its immediate playback. Whether we
are in management or in technical position this term
paper will help to understand the immense role of
audio compression.
I. INTRODUCTION
uncompressedThe trade-off between slightly reduced
audio quality and transmission or storage size is
outweighed by the latter for most practical audio
applications in which users may not perceive the loss
in playback rendition qualityFor example, one
Compact Disc holds approximately one hour of
uncompressed high fidelity music, less than 2 hours
of music compressed loss loss , or 7 hours of music
compressed in the MP3 format at medium bit rates.

Structured Audio means transmitting sound by

describing it rather than compressing it. That's the
whole idea, and it's a very simple one, but as you will
see if you keep reading, it leads to a wealth of new
directions for sound research and low bit-rate coding.
Audio compression is a form of data compression
designed to reduce the transmission bandwidth
requirement of digital audio streams and the storage
size of audio files. Audio compression algorithms are
implemented in computer software as audio codes.
Generic data compression algorithms perform poorly
with audio data, seldom reducing data size much
below 87% from the original, and are not designed
for use in real time applications. Consequently,
specifically optimized audio lossless and loss
algorithms have been created. Loss algorithms
provide greater compression rates and are used in Fig.1
mainstream consumer audio devices .In both loss and
lossless compression, information redundancy is II. HISTORY
reduced, using methods such as coding, pattern
recognition and linear prediction to reduce the
amount of information used to represent the
 A literature compendium for a large variety of audio
coding systems was published in the IEEE Journal on
Selected Areas in Communications (JSAC), February
1988. While there were some papers from before that
time, this collection documented an entire variety of
finished, working audio coders, nearly all of them
using perceptual (i.e. masking) techniques and some
kind of frequency analysis and back-end noiseless
coding. Several of these papers remarked on the
difficulty of obtaining good, clean digital audio for
research purposes. Most, if not all, of the authors in
the JSAC edition were also active in the MPEG-1
Audio committee.
The primary application areas of lossless encoding
are:
Archives
For archival purposes it is generally desired to
preserve the source material exactly (i.e. at
'best possible quality').
Editing
Audio engineers use lossless compression for
audio editing to avoid digital generation loss.
High fidelity playback
Audiophiles prefer lossless compression
form-ats to avoid compression artefacts.
Mastering of casual-use audio media
High quality master copies of recordings are
used to produce lossily compressed versions
for digital audio players. As formats and en-
coders improve, updated lossily compres-sed
files may be generated from the lossless
master.

Fig.2

Solidyne 922: The world's first commercial audio bit

compression card for PC, 1990

The world's first commercial broadcast automation

audio compression system was developed by Oscar
Bonello, an Engineering professor at the University
of Buenos Aires. In 1983, using the psychoacoustic
principle of the masking of critical bands first
published in 1967, he started developing a practical
application based on the recently developed IBM PC
computer, and the broadcast automation system was
launched in 1987 under the name Audio. 20 years
later, almost all the radio stations in the world were
using similar technology, manufactured by a number
of companies.

APPLICATIONS OF AUDIO
COMPRESION Fig.3
Compression Goals
 Reduced bandwidth
 Make decoded signal sound as close as
possible to original signal
 Lowest Implementation Complexity
 Robust
 Scalable

Fig.5

III. COMPRESION TECHNIQUE

 MPEG
 Voc File Compression
 Linear Predictive Coding
 Mu-law compression
 Differential Pulse Code Modulation

Fig.4

Sending Ethernet data, audio, video through power A. MPEG

line
* Tx / Rx Up to 4 optional RJ-45 ports
* Video Compression: MPEG-2
* Enhanced video quality by DBM B. Moving Picture Experts Group
* Video input format auto detect C. Part of a multiple standard for
* MPEG-1 / Layer 2 audio compression a. Video compression
* Typical 250ms latency time b. Audio compression
* Interference immunity for microwave oven,
WLAN, bluetooth, cordless phone
 c. Audio, Video and Data MPEG VEDIO
synchronization to an aggregate bit  Physically Lossy compression algorithm
rate of1.5 Mbit/sec.  Perceptually lossless, transparent algorithm
 Exploits perceptual properties of human ear
 Psychoacoustic modeling
MPEG Audio Standard ensures inter-operability,
defines coded bit stream syntax, defines decoding
process and guarantees decoder’s accuracy

B. VOC FILE COMPRESSION

The simplest compression techniques simply
removed any silence from the entire sample.
Creative Labs introduced this form of
compression with their introduction of the
Soundblaster line of sound cards. This
method analyzes the whole sample and then
codes the silence into the sample using byte
codes. It is very similar to run-length coding.
Linear Predictive Coding and Code Excited
Linear Predictor This was an early
development in audio compression that was
used primarily for speech. A Linear Predictive
Coding (LPC) encoder compares speech to an
analytical model of the vocal tract, then
throws away the speech and stores the
Fig.6 parameters of the best-fit model. The output
quality was poor and was often compared to
The Program Stream is similar to MPEG-1 computer speech and thus is not used much
Systems Multiplex today. A later development, Code Excited
Linear Predictor(CELP), increased the
complexity of the speech model further, while
allowing for greater compression due to faster
computers, and produced much better results.
Sound quality improved, while the
compression ratio increased. The algorithm
compares speech with an analytical model of
the vocal tract and computes the errors
between the original speech and the model. It
transmits both model parameters and a very
compressed representation of the errors.

C. MU-LAW AND A-LAW

COMPRESSION
Logarithmic compression is a good method because it
matches the way the human ear works. It only loses
information which the ear would not hear anyway,
and gives good quality results for both speech and
music. Although the compression ratio is not very
high it requires very little processing power to
achieve. It is the international standard telephony
encoding format, also known as ITU (formerly
CCITT) standard. It is commonly used in North
America and Japan for ISDN 8 kHz sampled, voice
grade, digital telephone service.
It packs each 16-bit sample into 8 bits by using a
logarithmic table to encode a 13-bit dynamic range,
dropping the least significant 3 bits of precision. The
quantization levels are dispersed unevely instead of
linearly to mimic the way that the human ear
perceives sound levels differently at different
frequencies. Unlike linear quantization, the
logarithmic step spacings represent low-amplitude
samples with greater accuracy than higher-amplitude
samples. This method is fast and compresses data
into half the size of the original sample. This method
is used quite widely due to the universal nature of its
adoption.
D. Differential Pulse Code Modulation
The ADPCM coder takes advantage of the fact that
neighboring audio samples are generally similar to
each other. Instead of representing each audio sample
independently as in PCM, an ADPCM encoder
computes the difference between each audio sample
and its predicted value and outputs thePCM value of
the differential. Note that the ADPCM encoder uses
most of the components of the ADPCM decoder
(Figure 2b) to compute the predicted values.
The quantizer output is generally only a (signed)
representation of the number of quantizer levels. The
re-quantizer reconstructs the value of the quantized
sample by multiplying the number of quantizer levels
by the quantizer step size and possibly adding an
offset of half a step size. Depending on the quantizer
implementation, this offset may be necessary to
center the re-quantized value between the
quantization thresholds.
The ADPCM coder can adapt to the characteristics of
the audio signal by changing the step size of either
the quantizer or the predictor, or by changing both.
The method of computing the predicted value and the
way the predictor and the quantizer adapt to the audio
signal vary among different ADPCM coding systems.
Some ADPCM systems require the encoder to
provide side information with the differential PCM
values. This side information can serve two purposes.
First, in some ADPCM schemes the decoder needs
the additional information to determine either the
predictor or the quantizer step size, or both. Second,
the data can provide redundant contextual
information to the decoder to enable recovery from
errors in the bit stream or to allow random access
entry into the coded bit stream.
The following section describes the ADPCM
algorithm proposed by the Interactive Multimedia
Association (IMA). This algorithm offers a
compression factor of (number of bits per source
sample)/4 to 1. Other ADPCM audio compression
schemes include the CCITT Recommendation G.721
(32 kilobits per second compressed data rate) and
Recommendation G.723 (24 kilobits per second
compressed data rate) standards and the compact disc
interactive audio compression algorithm.
The IMA ADPCM Algorithm. The IMA is a
consortium of computer hardware and software
vendors cooperating to develop a de facto standard
for computer multimedia data. The IMA’s goal for its
audio compression proposal was to select a public-
domain audio compression algorithm able to provide
good compressed audio quality with good data
compression performance. In addition, the algorithm
had to be simple enough to enable software-only,
real-time decompression of stereo, 44.1-kHz-
sampled, audio signals on a 20-megahertz (MHz)
386-class computer. The selected ADPCM algorithm
not only meets these goals, but is also simple enough
to enable software-only, real-time encoding on the
same computer.
The simplicity of the IMA ADPCM proposal lies in
the crudity of its predictor. The predicted value of the
audio sample is simply the decoded value of the
immediately previous audio sample. Thus the
predictor block in Figure 2 is merely a time-delay
element whose output is the input delayed by one
audio sample interval. Since this predictor is not
adaptive, side information is not necessary for the
reconstruction of the predictor.
Figure 3 shows a block diagram of the quantization
process used by the IMA algorithm. The quantizer
outputs four bits representing the signed magnitude
of the number of quantizer levels for each input
sample.
. model in the frequency domain, and latency is on the
Usability
Usability of lossy audio codecs is determined by:
Perceived audio quality Compression factor Speed of
compression and decompression Inherent latency of
algorithm (critical for real-time streaming
applications; see below) Product support Lossy
formats are often used for the distribution of
streaming audio, or interactive applications (such as
the coding of speech for digital transmission in cell
phone networks). In such applications, the data must
be decompressed as the data flows, rather than after
the entire data stream has been transmitted. Not all
audio codecs can be used for streaming applications,
and for such applications a codec designed to stream
data effectively will usually be chosen.
Latency results from the methods used to encode and
decode the data. Some codecs will analyze a longer
segment of the data to optimize efficiency, and then
code it in a manner that requires a larger segment of
data at one time in order to decode. (Often codecs
create segments called a "frame" to create discrete
data segments for encoding and decoding.) The
inherent latency of the coding algorithm can be
critical; for example, when there is two-way
transmission of data, such as with a telephone
conversation, significant delays may seriously
degrade the perceived quality.
In contrast to the speed of compression, which is
proportional to the number of operations required by
the algorithm, here latency refers to the number of
samples which must be analysed before a block of
audio is processed. In the minimum case, latency is 0
zero samples (e.g., if the coder/decoder simply
reduces the number of bits used to quantize the
signal). Time domain algorithms such as LPC also
often have low latencies, hence their popularity in
speech coding for telephony. In algorithms such as
MP3, however, a large number of samples have to be
analyzed in order to implement a psychoacoustic
order of 23 ms (46 ms for two-way communication).
 CONCLUSIONS
It has been shown that there is a wide range of
applications for audio compresion products.
Furthermore, ongoing research and development is
continually expanding the current range of
applications. one of the most important
characteristics of audio compresion is their clearity in
mpeg formet and to acquire data and making it small.

REFERENCES

[1].http://en.wikipedia.org/wiki/Audio_compressi
on_%28data%29.
[2].http://musik.ringofsaturn.com/compress.php
[3].www.ebookpdf.net/___A-Practical-Guide-To-
Video-and-Audio-
Compression.pdf_ebook_.html
[4].www.ics.uci.edu/~dan/class/267/notes/uci/z8.
pdf