Audio Engineering Society

Convention Paper 5987

Presented at the 115th Convention
2003 October 10–13 New York, NY, USA

This convention paper has been reproduced from the author’s advance manuscript, without editing, corrections,
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Society.

Lossless Compression for Audio Data in the

IEEE Floating-Point Format
Dai Yang1 , and Takehiro Moriya1
1
NTT Cyber Space Laboratory, 3-9-11, Midori-Cho, Musashio-Shi, Tokyo 180-8585, Japan

Correspondence should be addressed to Takehiro Moriya (t.moriya@ieee.org)

ABSTRACT

Most lossless audio-coding algorithms are designed for PCM input sound formats. Little work has been
done on the lossless compression of IEEE floating-point audio files. An efficient lossless-coding algorithm
that handles IEEE floating-point format data as well as PCM format data is described in this paper. In the
worst-case scenario, where the algorithm was applied to artificially generated 48-kHz sampling frequency and
32-bit floating-point sound files, an average compression ratio of 65% was still achieved for sound files with
48kHz sampling frequency. Moreover, the proposed algorithm is easily extensible to lossless/variable-lossy
operation, which will provide scalability to accommodate the requirements of a wider range of applications
and platforms.

1. INTRODUCTION of audio encoding, such as Dolby AC-3 and

ISO/MPEG audio standards, are capable of achiev-
Current state-of-the-art ”perceptual lossy” forms ing compression ratios up to 12:1 and higher and
YANG AND MORIYA FLOATING-POINT LOSSLESS AUDIO COMPRESSION
have thus become popular as components of internet
media applications, mobile terminals, and special-
purpose audio-visual data storage devices. However,
lossy audio-compression algorithms have met with
strong resistance from the fields of professional stu-
dio operations, sound archiving, and the disc-based
consumer market. This is because a lossy compres-
sion algorithm, regardless of how good it is, perma-
nently alters the original recorded sound data and
thus inherently reduces audio quality. In contrast,
copies of an audio waveform reproduced after lossless
audio compression are always bit-by-bit identical to
each other, no matter how many cycles of compres-
sion and decompression are applied.
In October of 2002, MPEG issued a new call for
proposals on MPEG-4 lossless audio coding [1, 2].
The new standard is now being developed and will be
finalized in a year or two. However, most of available
lossless audio coding algorithms focus on audio input
sources of the PCM format; little work has been done
on audio sources of the IEEE floating point format.
As an important sound format existing in the audio
industry, IEEE floating point sound files can store
an audio signal much more precisely than sound files
with regular PCM format can, and are, therefore,
much more difficult to be losslessly compressed.
In this paper, we describe a new lossless audio-
coding algorithm that is capable of handling input
sound files in the IEEE floating-point formats as well
as in various PCM formats. On the encoder side,
the input IEEE floating-point sound file is initially
converted into a PCM file. This file is then loss-
lessly compressed by an existing PCM lossless audio-
coding algorithm. Finally, the differences between
the original and PCM samples are obtained; this
data is then processed, compressed, and appended
to the bitstream.
The rest of this paper is organized as follows. Sec-
tion 2 describes the proposed algorithm in detail.
The experimental results and some discussion are
provided in section 3. Finally, some concluding re-
marks are offered in section 4.
2. COMPLETE ALGORITHM DESCRIPTION
2.1. System Overview
The proposed lossless audio-coding algorithm is ca-
pable of handling input sound files in the IEEE
AES 115TH CONVENTION, NEW YORK, NY, USA, 2003 OCTOBER 10–13
2
floating-point formats as well as in various PCM for-
mats. It has three major building blocks, i.e. for-
mat conversion, lossless PCM coding and difference
coding. In our implementation, Monkey’s audio is
adopted as the core coding module, i.e., for loss-
less PCM coding. The format-conversion module
handles the conversion of data between the IEEE
floating-point and PCM formats. The difference-
coding module, where the floating-point difference
data is precisely and efficiently coded, is the key to
the proposed algorithm’s fully lossless quality. In de-
signing the codec, extensive experiments were per-
formed to explore the trade-off between coding ef-
ficiency and complexity. On this basis, we selected
and applied the best combinations of parameter sets.
Fig. 1 illustrate the encoder block diagram. The in-
put IEEE floating-point sound file is initially con-
verted into PCM file M. This file is then loss-
lessly compressed by an existing PCM lossless audio-
coding algorithm. Finally, the differences between
the original and PCM samples are obtained; this
data is then processed, compressed, and appended
to the bitstream. The decoding process is just a
reverse procedure of the encoding.
2.2. Byte-wise Difference
Instead of doing a floating-point subtraction, the dif-
ference between the sample in IEEE floating-point
file A and those in file B are generated byte-wisely so
that the perfect reconstruction can always be guar-
anteed. To be more precise, we calculate the dif-
ference in the exponent D(e) and mantissa D(f ) for
each sample in files A and B. They can be calculated
either by bit-wise XOR or by a simple subtraction.
Each sample’s difference data is then stored in a 32-
bit format.
2.3. Format Conversion
How the PCM and floating-point format conversion
are handled has a great influence on the generation
of difference data, which affects the overall compres-
sion performance in extensive ways.
In most existing programs for wave file format con-
version [3], floating-point numbers are rounded to
the nearest integer when the IEEE floating-point
samples are converted to the PCM format. By do-
ing so, the magnitude of the residue is minimized.
However, the sign of each sample’s residue could be
YANG AND MORIYA FLOATING-POINT LOSSLESS AUDIO COMPRESSION

           !       

 ))  !  


    
 #% & %# %#
  % "!         !        ( '     



     #"        $"  

Fig. 1: Encoder block diagram.

$&%'! (& )'* +-,. /0) 2 4 3
= −1 1 • 7 J
2
       7 7
= −1 2 −23
• 7 0 7 6 0 = 111 6 00 6 6 6 0 5 2 = 2 • 1 6 1100 6 6 6 0 5
2
2
K
1
8 9 99: ; < => A E DB C ?@?@? H H
= −1 2 −23
•7 7 G I@I@I = 111 G I@I G G G I F 2 = 2 • 1 G 11I@I G G G I F
2
2

2
   "! #
          
= 1− 2 = 0 = 1 − 2 = 00

c −1
LMON&M P QR S T QU S VON-W S X Y Z Y U \0M&N _ b^ a ` ` ` ]] ` ` ` ]
2 −1
≤ <2 [ = 00 0

Fig. 2: An example of new format conversion.

positive or negative. Moreover, exponents of a cer- • Exponents remain unchanged, unless trunca-
tain amount of samples’ residue could be nonzero, tion to zero occurs
making the compression of residue data inefficient
and therefore degrading the overall coding perfor- • Signs of non-zero residues are always the same
mance. In the compression system described, the
magnitude of the floating-point number f is trun- • Number of non-zero bits in D(f ) can be calcu-
cated to the largest integer number that is less than lated implicitly
or equal to f , i.e. |f | → b|f |c. When comparing the
corresponding floating-point sample in IEEE files A The disadvantage of the new format conversion
and B, i.e. fA and fB , advantages and disadvantages method is that the magnitude of the residue is
of doing so over the traditional rounding method can greater than or equal to that generated by using the
be summarized as below: traditional rounding method.
Disadvantage: As for the advantages, it is obvious that fA is always
greater than or equal to fB , and fA (E) is always the
• The average magnitude of residue is increased same as fB (E) if no zero truncation occurs, where
fA (E) and fB (E) are the exponent of samples fA
Advantage: and fB .

AES 115TH CONVENTION, NEW YORK, NY, USA, 2003 OCTOBER 10–13
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YANG AND MORIYA FLOATING-POINT LOSSLESS AUDIO COMPRESSION
An example of how to calculate the number of
nonzero bits in D(f ) is shown in Fig. 2. Suppose
sample M in the intermediate 24-bit PCM file has
its magnitude equal to 7. When converted to the
IEEE 32-bit floating-point format, it has the form
of (−1)s 2−23 × 7.0, where 7.0 has the binary rep-
resentation of (111.000...0)2 . If the sample M is
converted by our new format conversion method, its
corresponding sample in IEEE file A should have the
form of (−1)s 2−23 × 7.xxx...x, where 7.xxx...x ≥ 7.0
and has the binary representation of (111.xxx...x)2 .
When byte-wise differences are performed, we notice
that the highest 2 bits in D(f ) are zeros. Given a
sample value in the intermediate PCM file, the num-
ber of nonzero bits in D(f ) can also be calculated
at the decoder side. Therefore, only those nonzero
bits of D(f ) need to be sent into the entropy coder
for further processing. On the contrary, if the tra-
ditional format conversion is performed, the number
of nonzero bits is not predictable, and all 23 bits of
D(f ) need to be passed to the entropy coder. This
simple example can be generalized into the following
rule.
Proposition If a sample in the 24-bit intermediate
PCM file has its magnitude greater than or equal
to 2n−1 , but is smaller than 2n (n > 0), then the
highest n − 1 bits in D(f ) should always be zero.
2.4. Intermediate PCM File
Most of the existing lossless PCM audio codecs sup-
port input sound files of 16- and 24-bit formats. And
in general, 16-bit PCM files can be more efficiently
compressed. As a major block in the compression
scheme described, lossless PCM coding part has a
strong influence on the overall coding performance.
Using a 16-bit PCM file as an intermediate PCM file
format will generate a much smaller sub-bitstream I,
while increasing residue values, when compared with
that of a 24-bit intermediate PCM file. Therefore,
a trade-off exists when choosing the precision of the
intermediate PCM file format.
2.5. Handling the Residue
The total bitstream is the combination of sub-
bitstreams I and D, where I is usually 15% to 30%
of the file size of the original IEEE 32-bit floating-
point file A. Intermediate raw difference data is of
the same size of the raw data in file A. Thus, achiev-
AES 115TH CONVENTION, NEW YORK, NY, USA, 2003 OCTOBER 10–13
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ing a satisfying compression result is impossible if
the residue data is handled poorly.
By using the new format conversion method, most
of samples’ D(e) equal zero. Therefore, these D(e)
do not need to be sent to the bitstream. As for these
samples’ corresponding D(f ) values, only those pos-
sible nonzero bits need to be buffered and sent to
an entropy coder. In the situation when a sample in
the original IEEE file A has its magnitude smaller
than 2−23 (or 2−15 ), zero truncation has to be done
when converting this number into a 24-bit (or 16-
bit) PCM format. Thus its corresponding D(e) is
nonzero and D(f ) may have all 23 nonzero bits.
By using our new format conversion method, the
number of highest zero bits of D(f ) can be calcu-
lated. However, the remaining lower bits of D(f )
may still contain several significant zero bits. In or-
der to further improve the performance, the highest
nonzero byte or bit in a whole block, Nhigh−block , is
calculated and sent to the bitstream. When com-
pressing D(f ), only the bits that are lower than
the smaller value of Nhigh−block and Nsample will be
processed, where Nsample is the number of possible
nonzero bits calculated by the new format conver-
sion method for each sample.
3. EXPERIMENTAL RESULTS
The proposed compression scheme was implemented
and tested in a Windows PC. We evaluated the per-
formance of the algorithm in the worst-case situation
by applying it to IEEE 32-bit floating-point format
audio source files. These files were artificially gen-
erated by multiplying PCM sound files using differ-
ent gain factors. Table 1 shows compression results
for two sets of seven input sound files with a sam-
pling frequency of 48-kHz. The number pair G/B in
the first column, e.g., 0.999/24, indicates the IEEE
floating-point input file is generated by multiplying
a gain of 0.999 to a 24-bit PCM file. And the num-
ber in column b, e.g., b=16, indicates the coding
results are obtained by using an intermediate PCM
file of 16-bit precision. From the figures in Table 1,
we found for input IEEE files A that are generated
from multiplying a gain to a 24-bit PCM file, the
coding performance when using a 24-bit intermedi-
ate PCM file is a little better than that for a 16-bit
YANG AND MORIYA FLOATING-POINT LOSSLESS AUDIO COMPRESSION

Table 1: Compression Ratio (48kHz)

G/B b Avemaria Cymbal Etude Flute Clarinet Haffner Violin Ave
0.999/24 24 62.30% 66.43% 62.95% 65.78% 65.89% 66.38% 67.84% 65.37%
16 62.43% 70.10% 63.03% 65.86% 65.91% 66.26% 67.84% 65.92%
1.001/24 24 62.30% 66.38% 62.94% 65.77% 65.89% 66.38% 67.84% 65.35%
16 62.43% 70.09% 63.03% 65.86% 65.91% 66.26% 67.84% 65.92%
1.0/24 24 43.93% 36.48% 45.57% 44.97% 48.43% 52.44% 49.07% 45.89%
16 44.26% 41.39% 45.84% 45.29% 48.63% 52.44% 49.27% 46.73%
0.97/16 24 62.37% 63.80% 63.01% 65.70% 65.93% 66.38% 67.82% 65.00%
16 61.02% 60.43% 61.31% 64.05% 64.63% 64.89% 66.57% 63.27%
1.001/16 24 62.47% 64.79% 63.07% 65.85% 65.97% 66.39% 67.87% 65.20%
16 61.63% 50.38% 62.43% 64.65% 65.44% 66.09% 67.16% 62.54%
1.0/16 24 44.10% 39.10% 45.69% 45.04% 48.51% 52.45% 49.10% 46.28%
16 19.14% 15.13% 20.71% 20.07% 23.48% 27.32% 24.07% 21.42%

intermediate PCM file case. However, these differ- 5. REFERENCES

ences are not signficant. For those input IEEE files
A that are generated from multiplying a gain to a
16-bit PCM file, the coding performance when using
a 24-bit intermediate PCM file is worse than that for
a 16-bit intermediate PCM file case. Also, consider-
ably large differences are observed in the case when
an input IEEE file A is a simple format conversion
of a 16-bit PCM file.
4. CONCLUSION
We described a new lossless compression scheme for
audio data in the IEEE 32-bit floating-point format.
The scheme is able to handle the lossless compression
of regular PCM input files as well. The algorithm
utilizes an existing PCM lossless audio coder as a
core coder, then gracefully handles the residue data
to generate the final bitstream. In the worst case
scenario, where the input IEEE floating-point file
is generated by multiplying a non-integer number
to a 24-bit PCM file, the proposed algorithm still
achieves an average compression ratio of about 65%
for input files with a sampling frequency of 48-kHz.
A compression ratio of 20% or less can be achieved
when the input IEEE floating-point file has the same
values as the 24- or 16-bit PCM file. Moreover, this
work is easily extensible to the handling of 64-bit
IEEE floating-point input source files, as well as to
a scalable lossy/lossless framework.
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proposals on MPEG-4 lossless audio coding, Kla-
genfurt, AT, July 2002.
[2] ISO/IEC JTC 1/SC29/WG11 N5208, Revised
call for proposals on MPEG-4 lossless audio cod-
ing, Shanghai, China, October 2002.
[3] P. Kabal, AFsp Library, programs
and routines, http://www.tnt.uni-
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[4] ANSI/IEEE Std. 754-1985 American Na-
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[5] T. Moriya, A. Jin, T. Mori, K. Ikeda, and
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AES 115TH CONVENTION, NEW YORK, NY, USA, 2003 OCTOBER 10–13
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