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MP3 AND AAC EXPLAINED
by KARLHEINZ BRANDENBURG1
MPEG AND INTERNET AUDIO
The proliferation of MPEG coded audio material on the
Internet has shown an exponential growth ”MP3” has
been featured in numerous articles in newspapers and periodicals
and on TV, mostly on the business pages because
of the potential impact on the recording industry.
While everybody is using MP3, not many (including
some of the software authors writing MP3 encoders,
decoders or associated tools) know the history and the details
of MPEG audio coding. This paper explains the basic
technology and some of the special features of MPEG-
1/2 Layer-3 (aka MP3). It also sheds some light on the
factors determining the quality of compressed audio and
what can be done wrong in MPEG encoding and decoding.
Why MPEG-1 Layer-3 ?
Looking for the reasons why MPEG-1/2 Layer-3 and not
other compression technology has emerged as the main
tool for Internet audio delivery, the following comes to
MPEG is defined as an open standard. The specification
is available (for a fee) to everybody interested
in implementing the standard. While there
are a number of patents covering MPEG Audio encoding
and decoding, all patent holders have declared
that they will license the patents on fair and
reasonable terms to everybody. No single company
”owns” the standard. Public example source code
is available to help implementers to avoid misunderstand
the standards text. The format is well
defined. With the exception of some incomplete
implementations no problems with interoperability
of equipment and software from different vendors
have been reported.
Availability of encoders and decoders
Driven first by the demand of professional use for
broadcasting, hardware (DSP) and software decoders
have been available for a number of years.
While audio compression is viewed as a main enabling
technology, the widespread use of computer
soundcards, computers getting fast enough to do
software audio decoding and even encoding, fast
Internet access for universities and businesses as
well as the spread of CD-ROM and CD-Audio
writers all contributed to the ease of distributing
music in MP3 format via computers.
In short, MPEG-1/2 Layer-3 was the right technology
available at the right time.
Newer audio compression technologies
MPEG-1 Layer-3 has been defined in 1991. Since then,
research on perceptual audio coding has progressed and
codecs with better compression efficiency became available.
Of these, MPEG-2 Advanced Audio Coding (AAC)
was developed as the successor for MPEG-1 Audio.
Other, proprietary audio compression systems have been
introduced with claims of higher performance. This paper
will just give a short look to AAC to explain the improvements
1. HIGH QUALITY AUDIO CODING
The basic task of a perceptual audio coding system is to
compress the digital audio data in a way that
the compression is as efficient as possible, i.e. the
compressed file is as small as possible and
the reconstructed (decoded) audio sounds exactly
(or as close as possible) to the original audio before
Other requirements for audio compression techniques include
low complexity (to enable software decoders or in-
Karlheinz Brandenburg MP3 and AAC explained
expensive hardware decoders with low power consumption)
and flexibility for different application scenarios.
The technique to do this is called perceptual encoding and
uses knowledge from psychoacoustics to reach the target
of efficient but inaudible compression. Perceptual encoding
is a lossy compression technique, i.e. the decoded file
is not a bit-exact replica of the original digital audio data.
Perceptual coders for high quality audio coding have been
a research topic since the late 70’s, with most activity occuring
since about 1986. For the purpose of this paper
we will concentrate on the format mostly used for Internet
audio and flash memory based portable audio devices,
MPEG-1/2 Layer-3 (aka MP3), and the format the author
believes will eventually be the successor of Layer-3,
namely MPEG-2 Advanced Audio Coding (AAC).
1.1. A basic perceptual audio coder
Fig 1 shows the basic block diagram of a perceptual encoding
Figure 1: Block diagram of a perceptual encoding/
It consists of the following building blocks:
A filter bank is used to decompose the input
signal into subsampled spectral components
(time/frequency domain). Together with the corresponding
filter bank in the decoder it forms an
Using either the time domain input signal and/or
the output of the analysis filter bank, an estimate of
the actual (time and frequency dependent)masking
threshold is computed using rules known from psychoacoustics.
This is called the perceptual model
of the perceptual encoding system.
Quantization and coding:
The spectral components are quantized and coded
with the aim of keeping the noise, which is introduced
by quantizing, below the masking threshold.
Depending on the algorithm, this step is done in
very different ways, from simple block companding
to analysis-by-synthesis systems using additional
Encoding of bitstream:
A bitstream formatter is used to assemble the bitstream,
which typically consists of the quantized
and coded spectral coefficients and some side information,
e.g. bit allocation information.
2. MPEG AUDIO CODING STANDARDS
MPEG (formally known as ISO/IEC JTC1/SC29/WG11,
mostly known by its nickname, Moving Pictures Experts
Group) has been set up by the ISO/IEC standardization
body in 1988 to develop generic (to be used for different
applications) standards for the coded representation
of moving pictures, associated audio, and their combination.
Since 1988 ISO/MPEG has been undertaking the standardization
of compression techniques for video and audio.
The original main topic of MPEG was video coding
together with audio coding for Digital Storage Media
(DSM). The audio coding standard developed by this
group has found its way into many different applications,
Digital Audio Broadcasting (EUREKA DAB,
WorldSpace, ARIB, DRM)
ISDN transmission for broadcast contribution and
Archival storage for broadcasting
Accompanying audio for digital TV (DVB, Video
Internet streaming (Microsoft Netshow, Apple
Portable audio (mpman, mplayer3, Rio, Lyra,
YEPP and others)
Storage and exchange of music files on computers
The most widely used audio compression formats are
MPEG-1/2 Audio Layers 2 and 3 (see below for the definition)
and Dolby AC-3. A large number of systems currently
under development plan to use MPEG-2 AAC.
MPEG-1 is the name for the first phase of MPEG work,
started in 1988, and was finalized with the adoption of
ISO/IEC IS 11172 in late 1992. The audio coding part
of MPEG-1 (ISO/IEC IS 11172-3, see  describes a
generic coding system, designed to fit the demands of
many applications. MPEG-1 audio consists of three operating
modes called layers with increasing complexity
Karlheinz Brandenburg MP3 and AAC explained
and performance from Layer-1 to Layer-3. Layer-3 is the
highest complexity mode, optimised to provide the highest
quality at low bit-rates (around 128 kbit/s for a stereo
MPEG-2 denotes the second phase of MPEG. It introduced
a lot of new concepts into MPEG video coding including
support for interlaced video signals. The main
application area for MPEG-2 is digital television.
The original (finalized in 1994) MPEG-2 Audio standard
 just consists of two extensions to MPEG-1:
Backwards compatible multichannel coding adds
the option of forward and backwards compatible
coding of multichannel signals including the 5.1
channel configuration known from cinema sound.
Coding at lower sampling frequencies adds sampling
frequencies of 16 kHz, 22.05 kHz and 24 kHz
to the sampling frequencies supported byMPEG-1.
This adds coding efficiency at very low bit-rates.
Both extensions do not introduce new coding algorithms
over MPEG-1 Audio. The multichannel extension contains
some new tools for joint coding techniques.
2.2.1. MPEG-2 Advanced Audio Coding
In verification tests in early 1994 it was shown that introducing
new coding algorithms and giving up backwards
compatibility to MPEG-1 promised a significant
improvement in coding efficiency (for the five channel
case). As a result, a new work item was defined and
led to the definition of MPEG-2 Advanced Audio Coding
(AAC) (, see the description in ). AAC is a second
generation audio coding scheme for generic coding
of stereo and multichannel signals.
The plan was to define the video coding for High Definition
Television applications in a further phase of MPEG,
to be called MPEG-3. However, early on it was decided
that the tools developed for MPEG-2 video coding
do contain everything needed for HDTV, so the development
for MPEG-3 was rolled into MPEG-2. Sometimes
MPEG-1/2 Layer-3 (MP3) is misnamed MPEG-3.
MPEG-4, finished in late 1998 (version 1 work, an
amendment is scheduled to be finished by end of 1999)
intends to become the next major standard in the world
of multimedia. Unlike MPEG-1 and MPEG-2, the emphasis
in MPEG-4 is on new functionalities rather than
better compression efficiency. Mobile as well as stationary
user terminals, database access, communications and
new types of interactive services will be major applications
for MPEG-4. The new standard will facilitate the
growing interaction and overlap between the hitherto separate
worlds of computing, electronic mass media (TV
and Radio) and telecommunications.
MPEG-4 audio consists of a family of audio coding algorithms
spanning the range from low bit-rate speech coding
(down to 2 kbit/s) up to high quality audio coding at
64 kbit/s per channel and above. Generic audio coding at
medium to high bit-rates is done by AAC.
Unlike MPEG-1/2/4, MPEG-7 does not define compression
algorithms. MPEG-7 (to be approved by July, 2001)
is a content representation standard for multimedia information
search, filtering, management and processing.
3. MPEG LAYER-3 AUDIO ENCODING
The following description of Layer-3 encoding focuses
on the basic functions and a number of details necessary
to understand the implications of encoding options on the
sound quality. It is not meant to be a complete description
of how to build an MPEG-1 Layer-3 encoder.
In order to be applicable to a number of very different application
scenarios, MPEG defined a data representation
including a number of options.
MPEG-1 audio works for both mono and stereo
signals. A technique called joint stereo coding can
be used to do more efficient combined coding of
the left and right channels of a stereophonic audio
signal. Layer-3 allows both mid/side stereo coding
and, for lower bit-rates, intensity stereo coding. Intensity
stereo coding allows for lower bit-rates but
brings the danger of a changing the sound image
(like moving instruments). The operating modes
– Single channel
– Dual channel (two independent channels, for
example containing different language versions
of the audio)
– Stereo (no joint stereo coding)
– Joint stereo
MPEG audio compression works on a number of
different sampling frequencies. MPEG-1 defines
audio compression at 32 kHz, 44.1 kHz and 48
kHz. MPEG-2 extends this to half the rates, i.e.
16 kHz, 22.05 and 24 kHz. MPEG-2.5 is the name
Karlheinz Brandenburg MP3 and AAC explained
Figure 2: Block diagram of an MPEG-1 Layer-3 encoder.
of a proprietary Fraunhofer extension to MPEG-
1/2 Layer-3 and works at 8 kHz, 11.05 and 12 kHz
MPEG audio does not work just at a fixed compression
ratio. The selection of the bit-rate of the compressed
audio is, within some limits, completely
left to the implementer or operator of an MPEG
audio coder. The standard defines a range of bitrates
from 32 kbit/s (in the case of MPEG-1) or
8 kbit/s (in the case of the MPEG-2 Low Sampling
Frequencies extension (LSF)) up to 320 kbit/s
(resp. 160 kbit/s for LSF). In the case of MPEG-
1/2 Layer-3, the switching of bit-rates from audio
frame to audio frame has to be supported by decoders.
This, together with the bit reservoir technology,
enables both variable bit-rate coding and
coding at any fixed bit-rate between the limits set
by the standard.
3.2. Normative versus informative
One, perhaps the most important property ofMPEG standards
is the principle of minimizing the amount of normative
elements in the standard. In the case of MPEG
audio this leads to the fact that only the data representation
(format of the compressed audio) and the decoder
are normative. Even the decoder is not specified in a bitexact
fashion but by giving formulas for most parts of the
algorithm and defining compliance by a maximum deviation
of the decoded signal from a reference decoder
implementing the formulas with double precision arithmetic
accuracy. This enables decoders running both on
floating point and fixed point architectures. Depending
on the skills of the implementers, fully compliant high
accuracy decoders can be done with down to 20 bit (in
the case of Layer-3) arithmetic wordlength without using
double precision calculations.
Encoding of MPEG audio is completely left to the implementer
of the standard. ISO/IEC IS 11172-3 (andMPEG-
2 audio, ISO/IEC 13818-3) contain the description of example
encoders. While these example descriptions have
been derived from the original encoders used for verification
tests, a lot of experience and knowledge is necessary
to implement good quality MPEG audio encoders. The
amount of investment necessary to engineer a high quality
MPEG audio encoder has kept the number of independently
developed encoder implementations very low.
3.3. Algorithm description
The following paragraphs describe the Layer-3 encoding
algorithm along the basic blocks of a perceptual encoder.
More details about Layer-3 can be found in  and .
Fig 2 shows the block diagram of a typical MPEG-1/2
The filterbank used in MPEG-1 Layer-3 belongs to the
class of hybrid filterbanks. It is built by cascading two
different kinds of filterbank: First the polyphase filterbank
(as used in Layer-1 and Layer2) and then an additionalModified
Discrete Cosine Transform (MDCT). The
polyphase filterbank has the purpose of making Layer-3
more similar to Layer-1 and Layer-2. The subdivision of
each polyphase frequency band into 18 finer subbands increases
the potential for redundancy removal, leading to
better coding efficiency for tonal signals. Another positive
result of better frequency resolution is the fact that
the error signal can be controlled to allow a finer tracking
of the masking threshold. The filter bank can be switched
Karlheinz Brandenburg MP3 and AAC explained
to less frequency resolution to avoid preechoes (see below).
3.3.2. Perceptual Model
The perceptual model is mainly determining the quality
of a given encoder implementation. A lot of additional
work has gone into this part of an encoder since the original
informative part in  has been written.
The perceptual model either uses a separate filterbank as
described in  or combines the calculation of energy
values (for the masking calculations) and the main filterbank.
The output of the perceptual model consists of values
for the masking threshold or allowed noise for each
coder partition. In Layer-3, these coder partitions are
roughly equivalent to the critical bands of human hearing.
If the quantization noise can be kept below the masking
threshold for each coder partition, then the compression
result should be indistinguishable from the original signal.
3.3.3. Quantization and Coding
A system of two nested iteration loops is the common solution
for quantization and coding in a Layer-3 encoder.
Quantization is done via a power-law quantizer. In this
way, larger values are automatically coded with less accuracy
and some noise shaping is already built into the
The quantized values are coded by Huffman coding. To
adapt the coding process to different local statistics of
the music signals the optimum Huffman table is selected
from a number of choices. The Huffman coding works
on pairs and, only in the case of very small numbers to be
coded, quadruples. To get even better adaption to signal
statistics, different Huffman code tables can be selected
for different parts of the spectrum.
Since Huffman coding is basically a variable code length
method and noise shaping has to be done to keep the
quantization noise below the masking threshold, a global
gain value (determining the quantization step size) and
scalefactors (determining noise shaping factors for each
scalefactor band) are applied before actual quantization.
The process to find the optimum gain and scalefactors
for a given block, bit-rate and output from the perceptual
model is usually done by two nested iteration loops in an
Inner iteration loop (rate loop)
The Huffman code tables assign shorter code
words to (more frequent) smaller quantized values.
If the number of bits resulting from the coding operation
exceeds the number of bits available to code
a given block of data, this can be corrected by adjusting
the global gain to result in a larger quantization
step size, leading to smaller quantized values.
This operation is repeated with different quantization
step sizes until the resulting bit demand
for Huffman coding is small enough. The loop
is called rate loop because it modifies the overall
coder rate until it is small enough.
Outer iteration loop (noise control loop)
To shape the quantization noise according to the
masking threshold, scalefactors are applied to each
scalefactor band. The systems starts with a default
factor of 1.0 for each band. If the quantization
noise in a given band is found to exceed
the masking threshold (allowed noise) as supplied
by the perceptual model, the scalefactor for this
band is adjusted to reduce the quantization noise.
Since achieving a smaller quantization noise requires
a larger number of quantization steps and
thus a higher bit-rate, the rate adjustment loop has
to be repeated every time newscalefactors are used.
In other words, the rate loop is nested within the
noise control loop. The outer (noise control) loop
is executed until the actual noise (computed from
the difference of the original spectral values minus
the quantized spectral values) is below the masking
threshold for every scalefactor band (i.e. critical
While the inner iteration loop always converges (if necessary,
by setting the quantization step size large enough to
zero out all spectral values), this is not true for the combination
of both iteration loops. If the perceptual model
requires quantization step sizes so small that the rate loop
always has to increase them to enable coding at the required
bit-rate, both can go on forever. To avoid this situation,
several conditions to stop the iterations early can
be checked. However, for fast encoding and good coding
results this condition should be avoided. This is one reason
why an MPEG Layer-3 encoder (the same is true for
AAC) usually needs tuning of perceptual model parameter
sets for each bit-rate.
4. MPEG-2 ADVANCED AUDIO CODING
Figure 3 shows a block diagram of an MPEG-2 AAC encoder.
AAC follows the same basic coding paradigm as Layer-3
(high frequency resolution filterbank, non-uniform quantization,
Huffman coding, iteration loop structure using
analysis-by-synthesis), but improves on Layer-3 in a lot
of details and uses new coding tools for improved quality
at low bit-rates.
4.1. Tools to enhance coding efficiency
The following changes compared to Layer-3 help to get
the same quality at lower bit-rates:
Higher frequency resolution
The number of frequency lines in AAC is up to
Karlheinz Brandenburg MP3 and AAC explained
Input time signal
Figure 3: Block diagram of an MPEG-2 AAC encoder.
1024 compared to 576 for Layer-3
An optional backward prediction, computed line
by line, achieves better coding efficiency especially
for very tone-like signals (e.g. pitchpipe). This feature
is only available within the rarely used main
Improved joint stereo coding
Compared to Layer-3, both the mid/side coding
and the intensity coding are more flexible, allowing
to apply them to reduce the bit-rate more frequently.
Improved Huffman coding
In AAC, coding by quadruples of frequency lines
is applied more often. In addition, the assignment
of Huffman code tables to coder partitions can be
much more flexible.
4.2. Tools to enhance audio quality
There are other improvements in AAC which help to retain
high quality for classes of very difficult signals.
Enhanced block switching
Instead of the hybrid (cascaded) filterbank in
Layer-3, AAC uses a standard switched MDCT
(Modified Discrete Cosine Transform) filterbank
with an impulse response (for short blocks) of 5.3
ms at 48 kHz sampling frequency. This compares
favourably with Layer-3 at 18.6 ms and reduces the
amount of pre-echo artifacts (see below for an explanation).
Temporal Noise Shaping, TNS
This technique does noise shaping in time domain
by doing an open loop prediction in the frequency
domain. TNS is a new technique which proves
to be especially successful for the improvement of
speech quality at low bit-rates.
With the sum of many small improvements,AAC reaches
on average the same quality as Layer-3 at about 70 % of
5. QUALITY CONSIDERATIONS
As explained above, the pure compliance of an encoder
with an MPEG audio standard does not guarantee any
quality of the compressed music. Audio quality differs
between different items, depending on basic parameters
including, of course, the bit-rate of the compressed audio
and the sophistication of different encoders even if they
work with the same set of basic parameters. To get more
insight about the level of quality possible with MP3 and
AAC, let us first have a look at typical artifacts associated
with perceptual audio coders.
5.1. Common types of artifacts
Unlike analog hi-fi equipment or FM broadcasting, perceptual
encoders when run at too low bit-rates or with the
wrong parameters exhibit sound deficiencies which are
in most cases different from the noise or distortion characteristics
we all are used to. The reason for this is the
Karlheinz Brandenburg MP3 and AAC explained
process generating differences in sound: The error introduced
by a high frequency resolution perceptual coder is
usually best modeled by a time-varying (in the rhythm of
the basic block or frame length) error at certain frequencies,
but not constrained to the harmonics of the music
So the signal may be sounding
distorted, but not like harmonic distortions.
noisy, but with the noise introduced only in a certain
rough, with the roughness often being very objectionable
because the error is changing characteristics
about every 20 ms.
5.1.1. Loss of bandwidth
If an encoder runs out of bits, i.e. it does not find a way
to encode a block of music data with the required fidelity
(e.g. allowed noise per critical band) within the bounds
of available bit-rate, some frequency lines might get set
to zero (deleted). The most common case is that some
high frequency content is lost. If the loss of bandwidth
is not constant, but changing from frame to frame (e.g.
every 24 ms) the effect becomes more objectionable than
in the case of a constant bandwidth reduction.
Preechoes are a very common and famous possible artifact
for high frequency resolution perceptual coding systems.
The name preecho is somewhat misleading: The
basic coding artifact is noise spread out over some time
even before the music event causing the noise. To understand
preechoes, let us have a look at the decoder of a
perceptual coding system (see Fig 1). The reconstructed
frequency lines are combined in the synthesis filterbank.
This filterbank consists of a modulation matrix and a synthesis
window. The quantization error in the coder can be
seen as a signal added to the original frequency line. The
length (in time) of such a signal is equal to the length
of the synthesis window. Thus, reconstruction errors are
spread over the full window length. If the music signal
contains a sudden increase in signal energy (like a castanet
attack), the quantization error is increased, too. As
explained above, this quantization error (noise) signal is
spread over the length of the synthesis window. If the attack
occured well within the analysis window, this error
will precede the actual cause for its existence in time. If
this prenoise extends beyond the premasking (backwards
masking) period, it becomes audible and is called preecho.
There are a number of techniques to avoid audible
preechoes (including variable bit-rate coding, local
increase in the bit-rate locally to reduce the amplitude of
the preecho), but overall this type of artifact belongs to
the most difficult to avoid.
0 8 16 24 32
Figure 4: Example of a pre-echo.
The lower curve (energy of the noise signal) shows the
form of the analysis window
5.1.3. Roughness, double-speak
Especially at low bit-rates and lower sampling frequencies
there is a mismatch between time resolution of the
coder (at least in its more efficient ”normal block” mode)
and the requirements to follow the time structure of some
signals. This effect is most noticeable for speech signals
and for listening via headphones. The effect is sometimes
called double-speak because a single voice tends
to sound like the same recording was done twice (with
the same speaker) and overlayed. AAC containes a technique
called Temporal Noise Shaping (TNS) which provides
noise shaping in time domain and thus virtually enhances
the time resolution of the filterbank.
5.2. Dynamic range and frequency response
There are some kinds of deficiencies of standard audio
equipment which cannot be found in properly designed
Layer-3 and AAC codecs. They are listed here to mention
the fact that it does not make sense to test for them. Most
MP3 and AAC both contain a global gain adjustment
parameter for every block of music data. According
to the word length and resolution of this
parameter, the dynamic range of both MP3 and
AAC is well beyond the equivalent of a 24 bit D/A
resolution. In short, MP3 and AAC represent the
Karlheinz Brandenburg MP3 and AAC explained
music in a way that the dynamic range of every
known audio source is perfectly retained.
Something similar is true for the frequency response
of an MP3 or AAC encoder/decoder pair.
Due to the nature of the coding process, the frequency
response (on average) is perfectly even (0
dB deviation). There may be short term variations,
but there is no coloration of the signal due to coding
artifacts (of course with the exception of parts
of the spectrum which are not transmitted at all).
5.3. Decoder differences
The decoders for both Layer-3 and AAC are fully specified
in the relevant ISO standards. The conformance
part of the standard gives the choice between standard
”MPEG-1 audio decoders” and ”high accuracy MPEG-1
audio decoders”, but up to now all implementations followed
the rules for high accuracy decoders. While the description
is not fully accurate to the bit (some parts are defined
via formulas allowing different methods of implementation),
it is done in a way that there can be no audible
differences between compliant decoders. This makes listening
tests of MP3 or AAC decoders a moot exercise:
The only question is whether a decoder is compliant to
the standard or not. One example for non-standard decoders
are implementations which do not allow switching
of the bit-rate within decoding a compressed bitstream.
The MPEG-1 standard specifically requires a decoder to
adapt to changing bit-rates. If a decoder is not able to
decode variable rate bitstreams, the decoder does not perform
properly. Another feature required in the standard
but not implemented in some (widely distributed) Layer-
3 decoders is the support of intensity stereo coding.
5.4. Not all encoders are created equal
The MPEG standards leave the implementation of an encoder
completely open. In the extreme case, one could
completely avoid implementing the perceptualmodel, decide
not to use the scalefactors (and therefore the outer
iteration loop) and do a very simple inner iteration loop.
Such an encoder would be very fast (potentially much
faster than any current encoder product), compliant with
the standard, produce nice audio quality for some signals
(where the build-in noise shaping of the non-uniform
quantizer is sufficient) but sound very bad for a large selection
of music. While this project is easy, it is much
more difficult to build an encoder with very high audio
quality across all types of music and even for the most
exotic test items. In MPEG, testing had always aimed
to verify sufficient encoder performance in worst case
scenarios. Nonetheless, the MP3 encoders around differ
quite a bit in their ability to produce, in a persistent way,
the best sounding compressed audio at low bit-rates.
5.4.1. Reference encoders
There are two sources for reference encoders:
The MPEG committee’s own software implementation
This encoder is the result of collaborative work of
a large number of individuals. The basic goal of
the effort was to provide a source for MPEG audio
bitstreams with correct syntax and to give an
implementation example helping people to understand
the syntax. High audio quality was not a
goal of the joint implementation effort. It can even
be said that some of the companies participating
in this effort are not interested to see a publicly
available high quality implementation of Layer-3
or AAC encoding. Therefore, encoders based on
the public source but without additional work on
encoding strategies and perceptual model usually
sound bad and should not be seen as examples for
”MP3 sound quality”.
Encoders used for the verification tests
In MPEG-1, verification tests used DSP based encoders.
MPEG-2 AAC development was based on
software simulation encoders. The source for the
encoders in all cases was the same development lab
which took the input from the MPEG Audio subgroup
and built the hardware or software encoders
for the tests. Only these encoders can be called
reference encoders in terms of encoding quality.
However, in all cases encoder development continues
well after the verifications tests leading to the
availability of improved encoders both from this
lab and others.
There are a number of tradeoffs in the design of an MP3
or AAC encoder. Up to now, the best quality has been
achieved using carefully tuned double iteration loop type
of encoders. This follows the paradigm as described
for the example encoder description in the MPEG standard.
These encoders are very slow. Faster, but maybe
somewhat lower quality encoders can be built by changing
the iteration strategy. Other differences between encoders
concern the psychoacoustic model, the strategy for
switching between the short and long window encoding
mode as well as the use of joint stereo coding. All this,
combined with the difficulties of tuning the encoder to
a given bit-rate and the choice of encoder bandwidth at
low bit-rates (see below) leads to quite some variation
between different encoders.
5.5. How to measure codec quality
To measure codec quality of high quality audio codecs
has, over the last ten years, developed to an art of its own.
Karlheinz Brandenburg MP3 and AAC explained
There are basically three measurement methods: Listening
tests, simple objectivemeasurementmethods and perceptual
5.5.1. Listening tests
To this date, large scale and well-controlled listening tests
are still the only method available to compare the performance
of different coding algorithms and different encoders.
The ITU-R (International Telecommunications
Union, Radiocommunications sector) with input from a
number of broadcasters and the MPEG audio group, has
developed a very elaborate set of rules for listening tests.
The aimof these tests is to stress the encoders underworst
case conditions , i.e. to find the material which is most
difficult to encode and to exhibit the performance of the
encoders under test for this material. This follows the observation
that in a lot of cases coding artifacts become
audible and even objectionable only after an extensive
training period. Since the use of equipment based on audio
compression technology (like memory based portable
audio players) itself constitutes extensive training, we can
expect that over time everybody becomes an expert listener.
Therefore, from the beginning, encoders should
better be tuned to satisfy the quality requirements of expert
The ITU-R test procedure requires a relatively large number
of test subjects and the test to be done in a double
blind fashion. More details about this type of testing can
be found in .
5.5.2. Simple objective measurement techniques
Over and over again people have tried to get a measure of
encoder quality by looking at units like Signal-to-Noise-
Ratio or bandwidth of the decoded signal. Since the basic
paradigm of perceptual coders relies on improving the
subjective quality by shaping the quantization noise over
frequency (and time), leading to an SNR lower than possible
without noise shaping, these measurements defy the
whole purpose of perceptual coding. The authors still using
these measurements just show that they have not understood
what they are doing. As explained below, to rely
on the bandwidth of the encoded signal does not show
much better understanding of the subject.
Another approach is to look at the codec result for certain
test signal inputs (transients, multi-tone signals). While
the results of such a test can tell a lot of information about
the codec to the expert, it is very dangerous to rely solely
on the results of such tests.
5.5.3. Perceptual measurement techniques
Beginning 15 years ago, there was a lot of research to apply
psychoacoustic modeling to the prediction of codec
quality and the audibility of certain artifacts. While the
state of the art is not yet sufficient to make large scale and
well-prepared listening tests obsolete, perceptual measurement
techniques have progressed to the point where
they are a very useful supplement to listening tests and
can replace them in some cases. ITU-R Task Group 10/4
worked for a number of years on the standardization of
perceptual measurement techniques and produced a recommendation
on a system called PEAQ (Perceptual Evaluation
of Audio Quality). The recommendation defines a
multi-mode system based on the collaborative effort of
all the leading laboratories working on perceptual measurement
techniques. For a more detailed description of
PEAQ, look for  in this conference proceedings.
5.6. Bit-rate versus quality
MPEG audio coding does not work with a fixed compression
rate. The user can choose the bit-rate and this
way the compression factor. Lower bit-rates will lead to
higher compression factors, but lower quality of the compressed
audio. Higher bit-rates lead to a lower probability
of signals with any audible artifacts. However, different
encoding algorithms do have ”sweet spots” where
they work best. At bit-rates much larger than this target
bit-rate the audio quality improves only very slowly
with bit-rate, atmuch lower bit-rates the quality decreases
very fast. The ”sweet spot” depends on codec characteristics
like the Huffman codebooks, so it is common to
express it in terms of bit per audio sample. For Layer-
3 this target bit-rate is around 1.33 bit/sample (i.e. 128
kbit/s for a stereo signal at 48 kHz), for AAC it is around
1 bit/sample (i.e. 96 kbit/s for a stereo signal at 48 kHz).
Due to the more flexible Huffman coding, AAC can keep
the basic coding efficiency up to higher bit-rates enabling
higher qualities. Multichannel coding, due to the joint
stereo coding techniques employed, is somewhat more
efficient per sample than stereo and again than mono coding.
A good overview of the tradeoff between bit-rates
and achievable quality for a number of coding algorithms
(including AAC and MP3) can be found in .
5.7. The bandwidth myth
Reports about encoder testing often include the mention
of the bandwidth of the compressed audio signal. In a
lot of cases this is due to misunderstandings about human
hearing on one hand and encoding strategies on the other
5.7.1. Hearing at high frequencies
It is certainly true that a large number of (especially
young) subjects are perfectly able to hear single sounds
at frequencies up to and sometimes well above 20 kHz.
However, contrary to popular belief, the author is not
aware of any scientific experiment which showed beyond
doubt that there is any listener (trained or not) able to detect
the difference between a (complex) musical signal
Karlheinz Brandenburg MP3 and AAC explained
with content up to 20 kHz and the same signal, but bandlimited
to around 16 kHz. To make it clear, there are
some hints to the fact that there are listeners with such
capabilities, but the full scientific proof has not yet been
given. As a corollary to this (for a lot of people unexpected)
theorem, it is a good encoding strategy to limit
the frequency response of an MP3 or AAC encoder to 16
kHz (or below if necessary). This is possible because
of the brick-wall characteristic of the filters in the encoder/
decoder filterbank. The generalization of this observation
to other types of audio equipment (in particular
analog) is not correct: Usually the frequency response of
the system is changed well below the cutoff point. Since
any deviation from the ideal straight line in frequency response
is very audible, normal audio equipment has to
support much higher frequencies in order to have the required
perfectly flat frequency response up to 16 kHz.
5.7.2. Encoding strategies
While loss of bandwidth below the frequency given by
the limits of human hearing is a coding artifact, it is not
necessarily the case that an encoder producing higher
bandwidth compressed audio sounds better. There is a
basic tradeoff where to spent the bits available for encoding.
If they are used to improve frequency response, they
are no longer available to produce a clean sound at lower
frequencies. To leave this tradeoff to the encoder algorithmoften
produces a bad sounding audio signalwith the
high frequency cutoff point varying from block to block.
According to the current state of the art, it is best to introduce
a fixed bandwidth limitation if the encoding is done
at a bit-rate where no consistent clean reproduction of the
full bandwidth signal is possible. Technically, both MP3
and AAC can reproduce signal content up to the limit
given by the actual sampling frequency. If there are encoders
with a fixed limited frequency response (at a given
bit-rate) compared to another encoder with much larger
bandwidth (at the same bit-rate), experience tells that in
most cases the encoder with the lower bandwidth produces
better sounding compressed audio. However, there
is a limit to this statement: At low bit-rates (64 kbit/s
for stereo and lower) the question of the best tradeoff in
terms of bandwidth versus cleanness is a hotly contested
question of taste. We have found that even trained listeners
sometimes completely disagree about the bandwidth
a given encoder should be run at.
5.8. Tuning for different bit-rates
As explained above, the double iteration loops do not
converge if there is a mismatch between the coding requirements
as given by the perceptual model and the bitrate
available to code a block of music. To avoid this situation,
it is wise to set the parameters in the psychoacoustic
model in a way that the iteration loops will normally
converge. This may require settings which lead to audible
differences, but the final coding result is still better than
the one from a perceptual model set to avoid any audible
difference combined with coding loops which do not converge
in a sensible way. To achieve this balance between
requirements from the perceptual model and bit-rate, the
coding parameters have to be readjusted if the encoder is
run at different bit-rates. This tuning procedure can be responsible
for a large share of the development effort for
an MP3 or AAC encoder.
6. FILE FORMATS
The MPEG standards define the representation of audio
data. Above this, MPEG as well defines how to put the
coded audio into a bitstream with synchronization and
header info sufficient to do the proper decoding without
any additional information given to the decoder.
6.1. MPEG-1/2 Layer-3 header format
MPEG-1/2 defines a mandatory header format which is
contained in every frame (every 24 ms at 48 kHz sampling
frequency). It contains, among others the following
Unlike in other standards, the sync word can occur
within the audio data, too. Therefore a proper
synchronization routine should check for the occurance
of more than one sync word in the right
distance and should completely resync only if no
more sync words are found at the right distance as
given by the bit-rate and sampling frequency.
The bit-rate is always given for the complete audio
stream and not per channel. In the case of Layer-3,
it is specifically allowed to switch the bit-rate on
the fly, leading to variable bit-rate encoding.
This will switch the decoder hardware (or software)
to different sampling frequencies, like 32
kHz, 44.1 kHz or 48 kHz in the case of MPEG-1.
The header contains information whether this is a
Layer-1, Layer-2 or Layer-3 bitstream (all share the
same header structure) and whether this is MPEG-
1 or MPEG-2 low sampling frequency encoding.
Again, as a fixed parameter this allows to differentiate
between mono, dual mono, stereo or joint
Karlheinz Brandenburg MP3 and AAC explained
Each header carries the two bits for the SCMS (Serial
Copy Management Scheme). However, since
the ease of manipulating these bits via software, the
practical importance of this way of copy protection
is relatively minor.
Due to the repetition of all information necessary to do a
successful decode in every frame, MPEG-1/2 bitstreams
are self sufficient and allow to start decoding at any point
in time. A properly built decoder even can read over other
information attached to the begin of an audio file (like
RIFF/WAV headers or metadata describing the content)
and then just start decoding the audio.
6.2. MPEG-2 AAC audio transport formats
While in MPEG-1 the basic audio format and the transport
syntax for synchronization and coding parameters
are tied together in an unseparable way, MPEG-2 AAC
defines both, but leaves the actual choice of audio transport
syntax to the application. The standard defines two
examples for the transport of audio data:
The ”Audio Data Interchange Format” puts all data
controlling the decoder (like sampling frequency,
mode etc.) into a single header preceeding the actual
audio stream. Thus it is useful for file exchange,
but does not allow for break-in or start of
decoding at any point in time like the MPEG-1 format.
The example ”Audio Data Transport Stream” format
packs AAC data into frames with headers very
similar to the MPEG-1/2 header format. AAC is
signaled as the (otherwise non-existant) ”Layer-4”
of MPEG Audio. Unlike Layer-3, the frame rate
is variable, containing always the audio data for
a complete frame between two occurances of the
sync word. ADTS again allows start of decoding
in the middle of an audio bitstream. The ADTS
format has emerged as the de-facto standard for a
number of applications using AAC.
Using an encoder with good performance, both Layer-
3 and MPEG-2 Advanced Audio Coding can compress
music while still maintaining near-CD or CD quality.
Among the two systems, Layer-3 at somewhat lower
complexity is the system of choice for current near-CD
quality applications. AAC is its designated successor,
providing near-CD quality at larger compression rates
(increasing the playing time of flash memory based devices
by nearly 50 % while maintaining the same quality
compared to Layer-3) and enabling higher quality encoding
and playback up to high definition audio (at 96 kHz
sampling rate). AAC, together with copyright protection
systems as defined by the Secure Digital Music Initiative
(SDMI), will be the compression system of choice
for future Electronic Music Distribution (EMD) and thus
follow in the footsteps of worldwide adoption of other
MPEG defined compression algorithms like MPEG Audio
Layer-2,MPEG Audio Layer-3 or MPEG Video.
The authorwould like to thank all collegues at Fraunhofer
IIS-A and MPEG Audio for all the wonderful collaborative
work over the last 11 years since the start of MPEG.
Special thanks are due to J¨urgen Herre, Harald Popp,
Martin Dietz, Oliver Kunz and J¨urgen Koller for helpful
suggestions. Part of the audio coding work at Fraunhofer
IIS-A has been supported by the Bavarian Ministry
for Economy, Transportation and Technology and the European
Commission (within the RACE and ACTS programmes).
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Karlheinz Brandenburg MP3 and AAC explained
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