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Presented at the 119th Convention
2005 October 710 New York, NY, USA
This convention paper has been reproduced from the authors advance manuscript, without editing, corrections, or
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ABSTRACT
MPEG-4 Audio Lossless Coding (ALS) is a new extension of the MPEG-4 audio coding family. The ALS
core codec is based on forward-adaptive linear prediction, which offers remarkable compression together with
low complexity. Additional features include long-term prediction, multichannel coding, and compression of
floating-point audio material. In this paper authors who have actively contributed to the standard describe
the basic elements of the ALS codec with a focus on prediction, entropy coding, and related tools. We also
present latest developments in the standardization process and point out the most important applications
of this new lossless audio format.
Multiplexing
Demultiplexing
Decoding
Long-Term Compressed
Prediction Bitstream
Long-Term
Prediction
Joint Channel
Coding
(Short-Term)
Prediction
Entropy
Coding
Block / Frame
Assembly Output
in section 2, section 3 puts the main focus on lin- coded. The encoder generates bitstream informa-
ear prediction together with block length switching, tion allowing for random access at intervals of sev-
random access, and long-term prediction. Section 4 eral frames. The encoder can also provide a CRC
illustrates methods for joint channel coding, and sec- checksum, which the decoder may use to verify the
tion 5 describes the entropy coding scheme for the decoded data.
prediction residual. In section 6, the codec exten-
sions for compression of floating-point audio data are 3. LINEAR PREDICTION
presented. Coding results for a variety of audio ma-
terial (including high-resolution and floating-point) Linear prediction is used in many applications for
are given in section 7, while section 8 provides a speech and audio signal processing. In the following,
discussion of application scenarios for lossless audio only FIR predictors are considered.
coding in general and MPEG-4 ALS in particular. 3.1. Prediction with FIR Filters
2. STRUCTURE OF THE CODEC The current sample of a time-discrete signal x(n) can
be approximately predicted from previous samples
The basic structure of the ALS encoder and decoder x(n k). The prediction is given by
is shown in Figure 1. The input audio data is par-
K
titioned into frames. Within a frame, each chan- X
nel can be further subdivided into blocks of audio x
(n) = hk x(n k), (1)
k=1
samples for further processing (block length switch-
ing). For each block, a prediction residual is calcu- where K is the order of the predictor. If the pre-
lated using forward adaptive prediction. The basic dicted samples are close to the original samples, the
(short-term) prediction can be combined with long- residual
term prediction. Inter-channel redundancy can be e(n) = x(n) x (n) (2)
removed by joint channel coding, either using differ-
ence coding of channel pairs or multi-channel coding. has a smaller variance than x(n) itself, hence e(n)
The remaining prediction residual is finally entropy can be encoded more efficiently.
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Liebchen et al. MPEG-4 Audio Lossless Coding
Encoder
Multiplexing
Predictor
Code Indices
Bitstream
Parcor
to LPC
Parcor Entropy
Q
Values Quantized Coding
Parcor Values
The procedure of estimating the predictor coeffi- encoder and the decoder, making the decoder signif-
cients from a segment of input samples, prior to fil- icantly more complex than in the forward-adaptive
tering that segment, is referred to as forward adap- case. MPEG-4 ALS specifies an optional backward-
tation. In that case, the coefficients have to be adaptive predictor as well [2], but in the follow-
transmitted. If the coefficients are estimated from ing, only the forward-adaptive predictor and related
previously processed segments or samples, e.g. from tools are discussed.
the residual, we speak of backward adaptation. This
3.2. Forward-Adaptive Prediction
procedure has the advantage that no transmission of
the coefficients is needed, since the data required to This chapter describes the forward-adaptive predic-
estimate the coefficients is available to the decoder tion scheme. A block diagram of the corresponding
as well [7]. encoder is shown in Figure 2.
Forward-adaptive prediction with orders around 10 The encoder consists of several building blocks. A
is widely used in speech coding, and can be employed buffer stores one block of input samples, and an ap-
for lossless audio coding as well [8][9]. The maxi- propriate set of parcor coefficients is calculated for
mum order of most forward-adaptive lossless predic- each block. The number of coefficients, i.e. the order
tion schemes is still rather small, e.g. K = 32 [10]. of the predictor, can be adapted as well. The quan-
An exception is the special 1-bit lossless codec for tized parcor values are entropy coded for transmis-
the Super Audio CD, which uses predictor orders of sion, and converted to LPC coefficients for the pre-
up to 128 [11]. diction filter, which calculates the prediction resid-
ual. The final entropy coding of the residual is de-
On the other hand, backward-adaptive FIR filters
scribed in section 5.
with some hundred coefficients are commonly used in
many areas, e.g. channel equalization and echo can- In forward-adaptive linear prediction, the optimal
cellation [12]. Most systems are based on the LMS predictor coefficients hk (in terms of a minimized
algorithm or a variation thereof, which has also been variance of the residual) are usually estimated for
proposed for lossless audio coding [13][14][15]. Such each block by the autocorrelation method or the co-
LMS-based coding schemes with high orders are ap- variance method [16]. The autocorrelation method,
plicable since the predictor coefficients do not have using the Levinson-Durbin algorithm, has the addi-
to be transmitted as side information, thus their tional advantage of providing a simple means to iter-
number does not contribute to the data rate. How- atively adapt the order of the predictor [8]. Further-
ever, backward-adaptive codecs have the drawback more, the algorithm inherently calculates the corre-
that the adaptation has to be carried out both in the sponding parcor coefficients as well.
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Liebchen et al. MPEG-4 Audio Lossless Coding
Decoder
Demultiplexing
Estimate
Bitstream Predictor
Code Indices
Parcor
Entropy Values Parcor
Decoder to LPC
The decoder (Figure 3) is significantly less complex with respect to the prediction order K. As the pre-
than the encoder, since no adaptation has to be car- diction gain rises monotonically with higher orders,
ried out. The transmitted parcor values are decoded, Re decreases with K. On the other hand Rc rises
converted to LPC coefficients, and are used by the monotonically with K, since an increasing number
inverse prediction filter to calculate the lossless re- of coefficients have to be transmitted.
construction signal. The computational effort of the
The search for the optimum order can be carried out
decoder mainly depends on the predictor orders cho-
efficiently by the Levinson-Durbin algorithm, which
sen by the encoder. Since the average order is typi-
determines recursively all predictors with increasing
cally well below the maximum order, prediction with
order. For each order, a complete set of predictor
greater maximum orders does not necessarily lead to
coefficients is calculated. Moreover, the variance e2
a significant increase of decoder complexity.
of the corresponding residual can be derived, result-
3.3. Adaptation of the Prediction Order ing in an estimate of the expected bit rate for the
Another crucial point in forward-adaptive prediction residual. Together with the bit rate for the coeffi-
is to determine a suitable prediction order. It is of cients, the total bit rate can be determined in each
course straightforward to use the same prediction iteration, i.e. for each prediction order. The opti-
order for all blocks of samples, thus adapting only mum order is found at the point where the total bit
the values of the coefficients. However, an adaptive rate no longer decreases.
choice of the number of predictor taps is extremely While it is obvious from Eq. (3) that the coefficient
beneficial in order to account for varying signal sta- bit rate has a direct effect on the total bit rate, a
tistics and different block lengths (see section 3.5), slower increase of Rc also allows to shift the mini-
as well as to minimize the amount of side informa- mum of Rtotal to higher orders (where Re is smaller
tion spent for transmitting the sets of coefficients. as well), which would lead to even better compres-
Assumed that the values of the coefficients are adap- sion. As MPEG-4 ALS supports prediction orders
tively chosen, increasing the order of the predictor up to K = 1023, efficient though accurate quantiza-
successively reduces the variance of the prediction tion of the predictor coefficients plays an important
error, and consequently leads to a smaller bit rate role in achieving maximum compression.
Re for the coded residual. On the other hand, the 3.4. Quantization of Predictor Coefficients
bit rate Rc for the predictor coefficients will rise with
the number of coefficients to be transmitted. Thus, Direct quantization of the predictor coefficients hk is
the task is to find the optimum order which mini- not very efficient for transmission, since even small
mizes the total bit rate. This can be expressed by quantization errors may result in large deviations
minimizing from the desired spectral characteristics of the opti-
mum prediction filter [7]. For this reason, the quan-
Rtotal (K) = Re (K) + Rc (K) (3) tization of predictor coefficients in MPEG-4 ALS is
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Liebchen et al. MPEG-4 Audio Lossless Coding
2000
1 -1
2 -1 -0.5 0 0.5 1
1500 3 r
4
frequency(r)
5
6 Fig. 5: Compander functions C(r) and C(r).
1000
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Liebchen et al. MPEG-4 Audio Lossless Coding
3.5. Block Length Switching is covered. For instance, a frame length of N = 8192
enables blocks with lengths NB = 8192, 4096, 2048,
The basic version of the encoder uses one sample
1024, 512, and 256.
block per channel in each frame. The frame length
can initially be adjusted to the sampling rate of the The choice of a suitable block partition is entirely
input signal, e.g. 2048 for 48 kHz or 4096 for 96 kHz left to the encoder, and thus not further specified
(approximately 43 ms in each case). by MPEG. Possible methods may range from eval-
uating of the signal statistics to exhaustive search
While the frame length is constant for one input file, algorithms. The actual partition has to be transmit-
optional block length switching enables a subdivision ted as side information, which takes at most 32 bits
of a frame into shorter blocks in order to adapt to per frame. Since the decoder still has to process the
transient segments of the audio signal. Previous ver- same number of samples per frame, block switching
sions of the MPEG-4 ALS codec [5] already included enables significantly improved compression without
a simple block switching mechanism which allowed increasing the decoder complexity.
to encode each frame of N samples using either one
full length block (NB = N ) or four blocks of length 3.6. Random Access
NB = N/4. Meanwhile, an improved and more flex- Random access stands for fast access to any part of
ible block switching scheme was designed, and each the encoded audio signal without costly decoding of
frame of length N can be hierarchically subdivided previous parts. It is an important feature for appli-
into up to 32 blocks. Arbitrary combinations of cations that employ seeking, editing, or streaming of
blocks with NB = N , N/2, N/4, N/8, N/16, and the compressed data.
N/32 are possible within a frame, as long as each In order to enable random access, the encoder has to
block results from a subdivision of a superordinate insert frames that can be decoded without decoding
block of double length. Therefore, a partition into previous frames. In those random access frames, no
N/4 + N/4 + N/2 is possible, whereas a partition samples from previous frames may be used for pre-
into N/4 + N/2 + N/4 is not (Figure 6). diction. The distance between random access frames
can be chosen from 255 to one frame. Depending on
N frame length and sampling rate, random access down
N/2 N/4 N/4
to some milliseconds is possible.
However, prediction at the beginning of random ac-
N/4 N/4 N/2 cess frames still constitutes a problem. A conven-
N/4 N/8 N/8 N/4 N/4 tional K-th order predictor would normally need K
samples from the previous frame in order the pre-
dict the current frames first sample. Since samples
N/4 N/2 N/4
from previous frames may not be used, the encoder
N/8 N/4 N/8 N/4 N/4 has either to assume zeros, or to transmit the first
K original samples directly, starting the prediction
at position K + 1.
Fig. 6: Block switching examples. The last two
partitions are not allowed due to the positions of As a result, compression at the beginning of ran-
the shaded blocks. dom access frames would be poor. In order to min-
imize this problem, the MPEG-4 ALS codec uses
Block length switching allows the use of both very progressive prediction [17], which makes use of as
short and very long blocks within the same audio many available samples as possible. While it is of
signal. For stationary segments, long blocks with course not feasible to predict the first sample of a
high predictor orders may be chosen, while for tran- random access frame, we can use first-order predic-
sient segments short blocks with lower orders are tion for the second sample, second-order prediction
more convenient. As the maximum block length is for the third sample, and so forth, until the samples
bounded by the frame length, the latter has to be from position K + 1 on are predicted using the full
chosen such that a reasonable range of block lengths K-th order predictor (Figure 7).
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Liebchen et al. MPEG-4 Audio Lossless Coding
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Liebchen et al. MPEG-4 Audio Lossless Coding
4.1. Difference Coding mode is independent coding, the second mode uses
3 taps, as shown in Figure 10, while the third mode
While it is straightforward to process two channels
uses 6 taps, additionally including 3 delayed taps as
x1 (n) and x2 (n) independently, a simple way to ex-
shown in Figure 11. At least one channel has to
ploit dependencies between these channels is to en-
be encoded in independent coding mode in order to
code the difference signal
decode all channels losslessly.
d(n) = x2 (n) x1 (n) (10)
For the 3-tap mode, the operation is performed ac-
instead of x1 (n) or x2 (n). Switching between x1 (n), cording to
x2 (n) and d(n) in each block can be carried out by !
1
comparison of the individual signals, depending on X
ec (n) = ec (n) j er (n + j) , (11)
which two signals can be coded most efficiently (see
j=1
Figure 9). Such prediction with switched difference
coding is beneficial in cases where two channels are
where ec (n) and ec (n) are residual signals of the cod-
very similar. In the case of multi-channel material,
ing channel, and er (n) is the residual of the reference
the channels can be rearranged by the encoder in
channel. The reference channel is searched among
order to assign suitable channel pairs.
available channels and the index is coded together
with multi-tap gain parameters j , which can be cal-
x1(n)
LPC EC culated by solving the normal equation
= X1 y, (12)
Comparison
LPC EC where
d(n)
T
= (1 , 0 , +1 )
r T r
er1 T er0 er1 T er+1
LPC EC e1 e1
x2(n) X = er1 T er0 er0 T er0 er0 T er+1
er1 T er+1 er0 T er+1 er+1 T er+1
Fig. 9: Switched difference coding (LPC - prediction, T
EC - entropy coding). y = ec T er1 , ec T er0 , ec T er+1
T
ec = (ec (0), ec (1), ec (2), , ec (N 1))
Besides simple difference coding, MPEG-4 ALS also
T
supports a more complex scheme for exploiting inter- er1 = (er (1), er (0), er (1), , er (N 2))
channel redundancy between arbitrary channels of er0 = (er (0), er (1), er (2), , er (N 1))
T
multi-channel signals, which is described in the fol- T
lowing section. er+1 = (er (1), er (2), er (3), , er (N ))
4.2. Multi-Channel Coding Here N is the number of samples per frame and eT
Lossless audio coding technology may be widely used denotes the transposed vector of e. The decoder
for compressing various multi-channel signals, such reconstructs the original residual signal by applying
as wave field synthesis signals, bio-medical and seis- simply the reverse operation:
mic signals as well as surround audio signals. !
1
X
To improve compression performance for these ec (n) = ec (n) + j er (n + j) (13)
multi-channel signals, adaptive subtraction from ref- j=1
erence channels with weighting factors is applied
based on inter-channel dependencies of the time do- The reconstructed residual signal ec (n) is used for
main prediction residual signal [18]. There are three short-term LPC synthesis or the preceding LTP de-
modes for each channel and each frame. The first coding process.
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Liebchen et al. MPEG-4 Audio Lossless Coding
Fig. 10: Three-tap subtraction. Fig. 11: Multi-tap subtraction with time difference.
For the 6-tap mode, adaptive subtraction is carried or the block can be further divided into four parts,
out using each encoded with a different Rice code. The in-
dices of the applied codes have to be transmitted, as
1
X shown in Figure 2. Since there are different ways to
ec (n) = ec (n) j er (n + j) determine the optimal Rice code for a given set of
j=1 data, it is up to the encoder to select suitable codes
1
!
X depending on the statistics of the residual.
r
+ +j e (n + + j) , (14)
j=1
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Liebchen et al. MPEG-4 Audio Lossless Coding
ever, to encode residuals in the center of the distrib- The integer sequence Y is compressed using the ba-
ution, the BGMC encoder splits them into LSB and sic ALS tools for integer data, while the difference
MSB components first, then it encodes MSBs using sequence Z is separately compressed by the masked
block Gilbert-Moore (arithmetic) codes, and finally Lempel-Ziv tool.
it transmits LSBs using direct fixed-lengths codes.
6.2. Estimation of the ACF
Both parameters emax and the number of directly
transmitted LSBs are selected such that they only Detecting the common multiplier A is not straight-
slightly affect the coding efficiency of this scheme, forward. So far, we have devised a reasonable esti-
while making it significantly less complex. mation procedure by analyzing the input signal X in
every frame, using a rational approximation of the
More detailed descriptions of the entropy coding
ACF. The estimation is similar to finding the great-
schemes used in MPEG-4 ALS are given in [3][19].
est common divisor with the condition that the in-
6. FLOATING-POINT AUDIO DATA put signal X may have an error at the unit in the
last position (ulp) due to rounding-off or chopping
In addition to integer audio signals, MPEG-4 ALS for truncation. The range of the error is between
also supports lossless compression of audio signals [-1/2, +1/2] of ulp. The relationship among the es-
in the IEEE 32-bit floating-point format [20]. The timated values of A, yi and xi can be expressed as
floating-point sequence is modeled by the sum of an
integer sequence and a residual sequence. The in- 1 1
xi ulpxi A yi xi + ulpxi , (17)
teger sequence is compressed using the basic ALS 2 2
tools for integer data, while the residual sequence is
where xi is the mantissa part of the ith floating-
compressed separately.
point sample in X, yi is the corresponding mantissa
6.1. Encoder for Floating-Point Audio Data part of ith floating-point sample in Y, and A is a
Figure 13 shows the integrated lossless encoder for normalized common multiplier. Rational approx-
integer and floating-point data. In the proposed en- imation using the continued fraction is applied to
coding scheme for 32-bit floating-point data, an in- estimate the ACF. The interval function for the ra-
put sequence X is decomposed into a common multi- tional approximation is
plier A, a multiplicand sequence Y, and a difference xi 1
ulpxi yi xi + 1
ulpxi
2 2
sequence Z. X and Z are vectors containing floating- 1 1 , (18)
x
+ ulpx yi
x ulpx
point values. Y is a sequence of truncated integers 2 2
and A is a scalar floating-point number. Thus, the where x is a selected sample of xi in the frame, and yi
input sequence can be written as is the estimated corresponding value of yi calculated
X = A Y + Z, (16) by x and ith sample xi . The common multiplier A
is normalized to 1.0 A < 2.0 since the mantissa
where is the multiplication with rounding. The bits of the floating-point data are also normalized to
rounding mode is set to round to nearest, to even 1.0 A < 2.0. The mantissa bits of y 1.5 and that
when tie. of y 3 are the same if the original mantissa bits of
First, the common multiplier A is estimated by ana- y are the same.
lyzing the input signal vector X in the current frame Once a reliable A is found, computing Y and Z from
using rational approximation of ACF (Approximate X and A is straightforward.
Common Factor). The common multiplier A is nor-
6.3. Masked-Lempel-Ziv Compression
malized to 1.0 A < 2.0. If an appropriate value of
ACF can not be found in the multiplier estimation Masked-LZ compression is one kind of dictionary-
module, the common multiplier A is set to 1.0. In based compression scheme. It is very similar to other
that case the input signal X is directly truncated to Lempel-Ziv compression variants, such as the LZW
integer Y, and the number of necessary bits for the compression scheme, in that there is a dictionary of
difference mantissa is uniquely determined by the previously encountered strings. The longest match
corresponding truncated integer y. string of input characters is searched using the string
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Liebchen et al. MPEG-4 Audio Lossless Coding
stored in the dictionary. The main difference is the An extensive description of compression techniques
way in which the input characters are compared with for floating-point audio data can be found in [21].
the characters of the string stored in the dictionary.
Masked-LZ uses a mask to compare them. The mask 7. COMPRESSION RESULTS
contains information about the bits of concern and In the following, different encoding modes of the
not of concern and uses it to compare two characters. MPEG-4 ALS reference codec [22] are compared in
6.4. Decoder for Floating-Point Audio Data terms of compression and complexity. The results
for several audio formats were determined for a low
For floating-point data, the integer multiplicand se-
complexity level (K 15, Rice Coding), a medium
quence Y is reconstructed and the multiplier A is
level (K 30, BGMC), and a maximum compres-
multiplied with it in order to obtain the floating-
sion level (K 1023, BGMC), all with random ac-
point sequence (Y A). The rounding mode round
cess of 500 ms. The results are also compared with
to nearest, to even when tie is used to round off the
the popular lossless audio codec FLAC [10] at max-
operation after the multiplication. The difference se-
imum compression (flac -8).
quence is decoded by the Masked-LZ decompression
module and converted to a floating-point format se- Apart from the bitstream syntax, MPEG does not
quence Z. If the multiplier A equals 1.0, the dif- specify how to realize some encoder features such
ference sequence is decoded using the word-length as predictor adaptation or block length switching.
information, which is defined from the value of the Even though we used the best ALS encoder imple-
corresponding integer value. Additional bits longer mentation so far, future improvements in terms of
than the necessary bit length are cut off (thrown compression, speed, and trade-off between those two
away) since they are dummy bits added by the en- are still possible.
coder. The tests were conducted on a 1.7 GHz Pentium-M
Both sequences, (Y A) and Z, are summed to gen- system, with 1024 MB of memory. The test material
erate the output floating-point sequence. The opera- was taken from the standard set of audio sequences
tion of the floating-point multiplication, truncation, for MPEG-4 Lossless Coding. It comprises nearly
and summation are emulated by integer multiplica- 1 GB of stereo waveform data with sampling rates
tion and summation in the decoding process. The of 48, 96, and 192 kHz, and resolutions of 16 and 24
integrated decoder is shown in Figure 14. bits.
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Liebchen et al. MPEG-4 Audio Lossless Coding
Unpack
Multiplier: A Num_bytes_diff_float
Floating-point Bitstream for
(A*Y)
output: X Diff_float_data
+ Masked-LZ
Differential decompression
signal: Z
7.1. Compression Ratio low complexity level is in any case less complex than
In the following, the compression ratio is defined as e.g. FLAC.
CompressedF ileSize The average CPU load for real-time decoding of var-
C= 100%, (19) ious audio formats, encoded at different complexity
OriginalF ileSize
levels, is shown in Table 2. Even for maximum com-
where smaller values mean better compression. The plexity, the CPU load of the MPEG-4 ALS refer-
results for the examined audio formats are shown in ence decoder is only around 20-25%, which in return
Table 1 (192 kHz material is not supported by the means that file based decoding is at least 4-5 times
FLAC codec). faster than real-time.
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Liebchen et al. MPEG-4 Audio Lossless Coding
further reduced by a factor of three (less than 2% Support for 32-bit floating-point audio data.
CPU load for 48 kHz material). Thus, MPEG-4
ALS data can be decoded even on hardware with Fast random access to the encoded data.
very low computing power. Optional storage in MPEG-4 file format (allows
7.3. Compression Ratio for Floating-Point Data multiplex with video and metadata).
Since currently there are only few tools avail- Examples for the use of lossless audio coding in gen-
able for lossless compression of floating-point audio, eral and MPEG-4 ALS in particular include both
MPEG-4 ALS is compared with WinZip [23], which professional and consumer applications:
is one of the most popular programs for dictionary
based lossless compression. Archival systems (broadcasting, studios, record
A set of 48 kHz test data for floating-point was gen- labels, libraries)
erated by weighted mixing of integer data using pro- Studio operations (storage, collaborative work-
fessional audio editing tools. The original integer in- ing, digital transfer)
put sequences sampled at 48 kHz in the stereo 24-bit
integer format were chosen from the test sets for High-resolution disc formats
MPEG-4 Lossless Coding again. A set of 96 kHz
test data, consisting of several original recordings, Internet distribution of audio files
was supplied by a professional sound studio. Online music stores (download)
The results in Table 3 show that ALS compresses Portable music players
floating-point audio data much more effectively than
a general-purpose compression tool such as WinZip. In the case online music stores, downloads of the lat-
est CD releases will no longer be restricted to lossy
Format Zip ALS formats such as MP3 or AAC. Instead, the consumer
medium
can purchase all tracks in full quality of the original
48 kHz / float 87.9 59.3 CD, but still receive the corresponding files at re-
96 kHz / float 89.5 47.8 duced data rates.
Table 3: Comparison of average compression ratios Furthermore, MPEG-4 ALS is not restricted to au-
for 32-bit floating-point data. dio signals, since it can also be used to compress
many other types of signals, such as medical (ECG,
EEG) or seismic data.
8. APPLICATIONS A global standard will facilitate interoperability be-
tween different hardware and software platforms,
MPEG-4 ALS defines a simple architecture of effi- thus promoting long-lasting multivendor support.
cient and fast lossless audio compression techniques
for both professional and consumer applications. It 9. CONCLUSION
offers many features not included in other lossless
compression schemes: MPEG-4 Audio Lossless Coding (ALS) is a highly
efficient and fast lossless audio compression scheme
for both professional and consumer applications
General support for virtually any uncompressed which offers many innovative features. It is based
digital audio format. on a codec developed by Technical University of
Berlin. Further improvements and extensions were
Support for PCM resolutions of up to 32-bit at contributed by RealNetworks and NTT.
arbitrary sampling rates.
Maximum compression can be achieved by means of
Multi-channel / multi-track support for up to high prediction orders together with efficient quan-
216 channels (including 5.1 surround). tization of the predictor coefficients and adaptive
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Liebchen et al. MPEG-4 Audio Lossless Coding
block length switching. Using low and medium com- [10] FLAC - Free Lossless Audio Codec,
plexity modes, real-time encoding and decoding is http://flac.sourceforge.net.
possible even on low-end devices.
[11] E. Janssen and D. Reefman, Super Audio
In the course of standardization, MPEG-4 ALS has CD: An Introduction, IEEE Signal Process-
reached the FDAM (Final Draft Amendment) stage ing Magazine, July 2003.
[2] in July 2005. It is therefore expected to become
[12] G.-O. Glentis, K. Berberidis, and S. Theodor-
an international standard by the end of 2005.
idis, Efficient Least Squares Adaptive Al-
gorithms For FIR Filtering, IEEE Signal
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