LOSSLESS AND LOSSY AUDIO DATA COMPRESSION
Kuncoro Triandono Mukti1 and Anggunmeka Luhur Prasasti2
Computer Engineering,
Telkom University
Bandung, Indonesia
1
kuncoroteem@gmail.com, 2 anggunmeka@gmail.com
Abstract - Audio is one of the fastest
growing multimedia data, especially in the
advanced music industry its make a lot of large
audio formats, such as data and video data, audio
data is also required for storage issues and real-
time access needs through computer networks.
The smaller audio size will reduce the delay time
so that data transmission will be faster.
Traditional data compression is applied to
computer machines, this is done because every
symbol that appears on the computer has
different bits. Data compression is used to
subtract the number of bits generated from each
symbol that appears. This compression is
expected to reduce (reduce the size of the data) in
the storage space. Similar to image compression,
there are two kinds of data compression
techniques on audio, namely lossy and lossless.
For daily consumption, lossy compression is
more widely used because the resulting
compression ratio is large or the resulting audio
size is very small. This paper discusses the basic
principles in the compression of audio,
Camparison Algortihm, and Audio file format.
Keywords - Audio Compression, Lossless,
Lossy.
I. INTRODUCTION
In general, data compression is a change of a
symbol into a code. Compression when there is a
very small code with the size of the original symbol
code. From a code or the basic symbols of a model
will be in a special code. Simply model a data set
and rules to set a symbol to determine a codecode as
the output [3].
Audio is one of the fastest growing
multimedia data, especially in the advanced music
industry its make a lot of large audio formats, such
as data and video data, audio data is also required for
storage issues and real-time access needs through
computer networks. The smaller audio size will
reduce the delay time so that data transmission will
be faster [1]. Like most compression techniques,
audio data compression, both lossy and lossless,
makes use of information redundancy with
encoding, pattern recognition, and linear predictions
like video compression. In lossless compression, the
compression result can be restored like the original
data without any change, then the compression ratio
cannot be too large to ensure all data can be restored
to its original form. Lossy compression is a
compression scheme that eliminates some of the
information contained in the original data so that
when done decompression process then the output
data will not exactly the same as the original data.
Both utilize the limitations of the human senses that
can only capture (perceive) environmental
conditions in a certain range, based on the frequency
of his voice is divided into 4 groups of which the
ears of human hearing is between 20 Hz to 20000
Hz.
Two of the most popular audio formats is
FLAC and MP3. In terms of quality, FLAC is
definitely better than MP3. However, in fact,
especially in Indonesia itself, more people choose to
listen to MP3 format (MPEG-1 Audio Layer 3) due
to its small size. If we compare with the MP3 format,
this FLAC format does require a large enough space.
If an audio CD-quality audio data use 44.1 kHz
sampling rate, 16 bits per sample, 2 channels
(stereo), then the total audio data storage per second
is approximately 176,400 Bytes so for a duration of
60 seconds (1 minute) it takes 10,584 MB. If the
average duration in a song is about 4 minutes, then
it takes about 42.336 MB of space to store 1 song
where 1 CD can hold only 16 songs [2, 4].
Currently, there are many compression
algorithms, including Dynamic Markov
Compression (DMC), Run Length Encoding (RLE),
Lempel Ziv Welch (LZW), Arithmetic coding,
Huffman Code, Rice Code, Golomb Code, BW
Transform, and others. Based on some previous
research the Huffman algorithm is faster at
compression, and better in audio compression,
according to [11] Huffman algorithm is better,
faster, and produces high PSNR than Arithmetic
coding in compression. And according to [12],
Huffman compression results better than LZW and
DMC in the case of binary files, multimedia files,
image files and compressed files.
In this paper first we will explain the basic
theory of lossless and lossy compression, then the
algorithm that can be used to compress, and then
compare the results of previous research.
II. LOSSLESS & LOSSY COMPRESSION
Data compression is the process of encoding
information using bits or other information-bearing
units that are lower than the data representation that
is not encoded with a particular encoding system,
there are two compression techniques Lossless and
Lossy. Here is an explanation of Lossless, Lossy and
audio data compression algorithms. For specific
classification shown in figure 1.
figure 1. Classification data compression
2.1. LOSSLESS
Lossless compression in audio data means that
the compressed results of the data can be
decompressed to produce exactly the same data as
the original data, without any loss of quality at all.
Lossless compression for audio data is somewhat
similar to the generic lossless compression
algorithm, with a compression ratio of about 50% to
60%, although it can achieve 35% in orchestral
music data or less noise chorus.
usually a lossless compression program uses
two different types of algorithms. First, which
produces statistical models for data input, and
second, maping the input data to the bit sequence
using this model in the way that "probable" data will
produce shorter outputs of "improbable" data. The
main encoding algorithms used to generate bit
sequences are huffman coding and arithmethic
coding [6].
figure 2. Lossless compression
A. Usability of Lossless Compression
Lossless compression is primarily used
for archiving, and editing. For the purposes of
archiving, of course, the desired quality is the
best quality. So also with editing. Editing lossy
compressed data causes a decrease in the sound
quality of each storage. Then lossless
compression is always used in sound
engineering. In addition to both uses, lossless
compression is also commonly used by
audiophile, the music fans who enjoy listening
to music with high quality with high-quality
hardware as well. Lossless compressed audio
data is also used to generate lossy audio data for
distribution. Nowadays, with the increasingly
low cost of digital data storage media and
bandwidth, lossless compression is becoming
increasingly popular among consumers.
B. Basic Principle of Lossless Compression
There are two main stages in lossless
compression for audio data, first prediction, and
coding. Prediction uses the previous samples to
predict the next sample. Then the difference
between the sample predicted the result and the
actual sample is coded. For each format,
usually, the difference is only found in
prediction and/or coding techniques.
Some audio formats that support lossless
compression include Shorten, while those
commonly used today are Free Lossless Audio
Codec (FLAC), Apple Lossless, MPEG-4 ALS,
Monkey's Audio, WavPack, and True Audio.
Each format has a different step or stage on its
compression as an example in this paper will
explain the compression process in FLAC
format.
Example Audio Compression on FLAC
The FLAC format is issued by the
Xiph.Org Foundation by utilizing the high
correlation between samples in the audio data.
FLAC uses linear predictions to convert
samples into rows of numbers called residues,
which are then stored with Golomb-Rice
coding. The resulting compression ratio is 40%
to 50%.
The compression process is done by
several stages:
 Blocking, Block in FLAC refers to a row of
samples spanning multiple channels. Block
size may vary depending on several factors
including sample rate. This block size
affects the compression ratio directly. If the
block size is too small, it takes a lot of
frames so many bits will be wasted to store
the frame header. If it is too large, the
characteristic of the audio signal will be too
varied making it difficult to find an optimal
predictor. FLAC limits block size between
16 to 65535 samples per block.
 Interchannel Decorrelation, For stereo data,
there are often many correlations between
the left and right channels. Thus there are
several channel storage methods into
blocks as follows:
o Independent, Independent, both
channels are encoded separately.
o Mid-side, storing the average signal
of both channels as mid channel and
the difference between the left
channel and the right as the side
channel.
o Left-side, save left channel and side
channel.
o Right-side, save right channel and
side channel.
In certain cases, Left and Right is the most
efficient method [15].
 Prediction, the encoder looks for an
approximation of the mathematical
description of the signal on each block. The
size of this description is generally much
smaller than the size of the signal itself.
This prediction method is known by
encoder or decoder so that the compression
results enough to include the prediction
parameters. There are four methods used
by FLAC for prediction:
o Verbatim, The prediction signal is
zero, so the residue is the same as the
actual signal (no compression).
o Constant, This method is used if in the
certain channel in a block there is
digital silence or constant value. The
encoding used is run-length.
o Fixed linear prediction
o FIR linear prediction
 Residual coding using Golomb-Rice
Coding, when predictors cannot describe
signals exactly, it is necessary to keep the
difference between the original signal and
the predicted signal. This difference is
called a residue. The effectiveness of the
prediction can be seen from the size of the
required residue. This residue is stored with
one of two ways of rice coding:
o Using one parameter for the entire
residue. This parameter is based on
the residual value variance.
o The residue is divided into several
parts of the same length, with each
section having its own parameters
determined from the average value of
the residue.
C. Algorithm for Lossless Compression
 Huffman Code
 Golomb Code
 Rice Code
 Tunstall Code
 Arithmetics Code
 Dictionary Code
 Run-Length Code
2.2. LOSSY
The compression technique where the
decompression data is not the same as the data
before compression but is "enough" to be used.
Examples: Mp3, streaming media, JPEG, MPEG,
and WMA. The advantage of this method is its
smaller size compared to Lossless. In the lossy
compression for audio data, there will be a decrease
in quality if the compression results are attempted to
be decompressed. This quality degradation is called
compression “artifacts”.
 Usually this technique removes pieces of data
that are actually not so useful, not so perceived, not
so visible, so people still think that the data can still
be used even if it is compressed. for example on
MP3. examples of these methods are Transform
Coding, Wavelet, and others. lossy compression is
also called irreversible compression because the
original data is impossible to restore. the advantages
of this technique is a high compression ratio
compared to lossless methods [6].
figure 3. Lossy Compression
A. Usability of Lossy Compression
Lossy compression in audio data is very
widely used, either directly (eg on mp3 players)
or indirectly (contained in DVD video, digital
television, video streaming, etc.). This
compression is used by the music lovers
because the result of a very high compression
ratio between 0% to 100% with sound quality is
still "good enough".
B. Basic Principle of Lossy Compression
Primarily, lossy compression of audio data
utilizes psychoacoustic, which is related to the
ability of the human ear to sound. The human
ear can only capture sounds with frequencies
between 20Hz and 20000Hz. But there are still
some other basic techniques in lossy
compression for audio data, namely:
 Voc File Compression. This technique
is very simple, ie removing samples of
silent samples (no sound) such as
pauses between paragraphs in a speech
or a moment's silence on some parts of
a song.
 Linear Predictive Coding (LPC), Code
Excited Linear Predictor (CELP).
 CELP is a further development, with a
more complex analytical model to
produce greater compression ratios
and better sound quality. Slightly
similar to lossless compression, at
CELP the difference between the
original sound and the analytical
model is also stored in a compressed
form as well.
 ADPCM, Simply the first sample is
kept intact, while for the next samples,
the saved is the difference with the
previous sample, which is generally
not very large.
 MPEG, This technique uses
psychoacoustic theory. If the voice
cannot be heard by the human ear, then
the sound part does not need to be
encoded. In addition, which is still
associated with psychoacoustic is
noise shaping. High-frequency signals
can only be heard by humans if they
have large volumes, therefore noise is
'hidden' in these high-frequency areas
with small volumes.
Some audio formats that support lossy
compression include a very popular format
is MP3 which is part of MPEG that handles
the audio layer (MPEG layer III), AAC
which is further developed, and OGG. For
speech data, there are several formats such
as A-law / μ-law used on the phone, AMR
on GSM, AMR WB for CDMA, and so on.
C. Algorithm for Lossy Compression
 Scalar Quantization
 Vector Quantization
III. AUDIO FILE FORMAT
Audio file format grouped in 3 classification
which is Free and Open file format, Open File
Format and Proprietary Format. More detailed
shown on table 1.
Table 1. Audio File Format [16]
Free and Open File format
Format Descripton
standard audio file
container format used
mainly in Windows PCs.
Commonly used for storing
uncompressed (PCM), CD-
quality sound files, which
*.wav
means that they can be large
in size — around 10 MB per
minute. Wave files can also
contain data encoded with a
variety of codecs to reduce
the file size (for example the
 GSM or mp3 codecs). Wav an industry-standard
files use a RIFF structure. protocol that enables
a free, open source electronic musical
container format supporting instruments, computers, and
*.mid
a variety of codecs, the most other equipment to
*.ogg popular of which is the communicate, control, and
audio codec Vorbis. Vorbis synchronize with each
offers compression similar other.
to MP3 but is less popular.
Open File Format
Musepack or MPC
(formerly known as Format Description
MPEGplus, MPEG+ or
MP+) is an open source designed for telephony use
lossy audio codec, in Europe, gsm is a very
specifically optimized for practical format for
transparent compression of telephone quality voice. It
stereo audio at bitrates of *.gsm makes a good compromise
160–180 kbit/s. Musepack between file size and
*.mpc and Ogg Vorbis are rated as quality. Note that wav files
the two best available can also be encoded with
codecs for high-quality the gsm codec.
lossy audio compression in A variable codec format
many doubleblind listening designed for dictation. It has
tests. Nevertheless, dictation header
Musepack is even less *.dct information and can be
popular than Ogg Vorbis encrypted (often required
and nowadays is used by medical confidentiality
mainly by the audiophiles. laws).
a lossless compression the vox format most
codec. This format is a commonly uses the
lossless compression as like Dialogic ADPCM
zip but for audio. If you (Adaptive Differential
compress a PCM file to flac Pulse Code Modulation)
and then restore it again it codec. Similar to other
will be a perfect copy of the ADPCM formats, it
original. (All the other compresses to 4-bits. Vox
*.vox
codecs discussed here are format files are similar to
*.flac wave files except that the
lossy which means a small
part of the quality is lost). vox files contain no
The cost of this losslessness information about the file
is that the compression ratio itself so the codec sample
is not good. Flac is rate and number of channels
recommended for archiving must first be specified in
PCM files where quality is order to play a vox file.
important (e.g. broadcast or the Advanced Audio
music use). Coding format is based on
the standard audio file the MPEG2 and MPEG4
*.aac
*.aiff format used by Apple. It is standards. aac files are
like a wav file for the Mac. usually ADTS or ADIF
a raw file can contain audio containers.
in any codec but is usually MPEG-4 audio most often
*. raw used with PCM audio data. *.mp4/m4a AAC but sometimes
It is rarely used except for MP2/MP3.
technical tests. a Samsung audio format
the standard audio file *.mmf that play a music of
format used by Sun, Unix ringtone.
and Java. The audio in au Proprietary Formats
*.au
files can be PCM or
compressed with the μ-law, Format Description
a μlaw or G729 codecs.
 the MPEG Layer-3 format A proprietary version with
is the most popular format Digital Rights Management
for downloading and storing developed by 3D Solar UK
music. By eliminating Ltd for use in music
*.ivs
portions of the audio file downloaded from their
that are essentially Tronme Music Store and
*.mp3
inaudible, mp3 files are interactive music and video
compressed to roughly player.
onetenth the size of an A proprietary version of
equivalent PCM file while AAC in MP4 with Digital
maintaining good audio Rights Management
*.mp4
quality. developed by Apple for use
the popular Windows in music downloaded from
Media Audio format owned their iTunes Music Store.
by Microsoft. Designed An iKlax Media proprietary
*.wma
with Digital Rights format, the iKlax format is a
Management (DRM) multi-track digital audio
abilities for copy protection *.iklax format allowing various
the older style Sony actions on musical data, for
ATRAC format. It always instance on mixing and
*.wav has a .wav file extension. To volumes arrangements.
open these files simply a Musinaut proprietary
install the ATRAC3 drivers. format allowing play of
*.mxp4
a Real Audio format different versions (or skins)
designed for streaming of the same song.
audio over the Internet. The
.ra format allows files to be For example compression with different format
*.ra stored in a self-contained shown in figure 4.
fashion on a computer, with
all of the audio data
contained inside the file
itself.
a text file that contains a
link to the Internet address
where the Real Audio file is
*.ram
stored. The .ram file
contains no audio data
itself.
Digital Speech Standard
files are an Olympus
proprietary format. It is a
fairly old and poor codec.
*.dss
Prefer gsm or mp3 where
the recorder allows. It
allows additional data to be
held in the file header.
a Sony proprietary format
*.msv for Memory Stick
compressed voice files.
a Sony proprietary format
for compressed voice files;
*.dvf
commonly used by Sony
dictation recorders. figure 4. Compression with different file format
 IV. RESULTS AND DISCUSSION

Table 2. Comparison of Algorithm Lossy and Lossless

No. Format Compression Results

Algorithm Type
Referensi (%)
Source Results
*.aac *.aac 3%
*.flac *.flac 1%
[1] Huffman Lossless *.midi *.midi 1% - 35%
*.mp3 *.mp3 1% - 3%
*.wav *.wav 10% - 27%
Huffman
[2] Lossy *.flac *.mp3 90% - 95%
Shift Coding
*.mp3 *.mp3 1%
[3] Huffman Lossless *.wav *.wav 17%
*.wma *.wma 2%

[4] Huffman Lossless *.wav *.wav 20% - 40%

*.mp3 *.mp3 2.4%
[5] Arithmethic Lossy
*.wav *.wav 11%
*.aif *.aic 18.443%
*.au *.aic 27.589%
[6] Arithmethic Lossy
*.mid *.aic 31.819%
*.wav *.aic 36.741%
Huffman - - 40.56%
[7] Lempel-Ziv Lossless
- - 51.47%
Welch (LZW)
Run Length *.mp3 *.mp3 0.46 %
[8] Lossless
Encoding (RLE) *.wav *.wav 13.83 %
Lempel-Ziv
[9] Lossless *.mp3 *.wav 3.73% - 29,56%
Welch (LZW)
Huffman Shift
[10] Lossless *.wav *.wav 14,87%
Coding

In the study [1], the compression and V. CONCLUSION

decompression testing on 30 sample data to see the
results of the ratio and time, after the test obtained
the conclusion if the highest to lowest compression
ratio in the test data format can be sorted as follows
*.midi, *.wav , *.mp3, *.aac, *.flac. In contrast to
the research [1], [3], [4], [7] that using only the
Huffman algorithm, the research [2] used the
Huffman Shift Coding algorithm, with the aim of
performing a lossy compression obtained with a
very satisfactory compression ratio reaching 90%.
Then in the study [5], [6] using Arithmetic Coding
to perform compression Lossy only get the highest
compression ratio of 31.819% in *.wav format.
while in research [7], [9] the algorithm used is
Lempel-Ziv Welch (LZW) with lossless
compression. The highest compression ratio was
51.47%. and of all the proposed algorithms
compared to Run Length Encoding (RLE) in the
study [8] only got a 0.46% compression ratio alone.
From previous research, can be concluded if
currently the popular algorithm that is Huffman
Coding for Lossless compression, with the
advancement of data storage technology today a
variety of lossless compression began to be
developed due to the storage price which can be
spelled has started relatively cheap and to get
comfortable while listening to music.
From the various formats used as an object,
testing got the conclusion if the format * .wav is the
format that has the best compression results
compared with other formats.
If grouped by type and algorithm, Lossless
compression is best suited to use Lempel-Ziv Welch
(LZW) which has been proven by research [7], and
for Lossy is Huffman Shift Coding algorithm that
achieves a compression ratio of over 90% [7].
At least, that all based on the previous research
for now all compression application or application
that using audio compression such as Factory
Format, Xilisoft, Instagram, Facebook, Line,
Whatsapp they use their own compression that
produces different results even though the size of the
data is not too much different. therefore the
compression percentage is not very important
because the percentage of compression can be
adjusted but what is important now is a good
compression result. for example by using MSE
(mean square error) we can know the difference of
the original file with the compression result, the
smaller value is good compression results. If the
compression reaches 100% but the value of MSE is
very large the compression results would not be
good.
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