AOH :: MP3.FAQ|
Frequently Asked Questions about MPEG-3 compression.
Frequently Asked Questions about MPEG Audio Layer-3, Fraunhofer-IIS, and
all the rest...
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Table of Contents
* Introduction - or: What is "MPEG Audio Layer-3"?
* Applications - or: Layer-3, what is it good for?
* Overview about the ISO-MPEG Standard - or: What is MPEG all about?
* Some Basics about MPEG Audio - or: What about Layer-1, Layer-2,
* Advanced Features of Layer-3 - or: Why does Layer-3 perform so
* Basics of Perceptual Audio Coding - or: What is the trick?
* References - or: Where to find more information?
* About us - or: What is going on at our Fraunhofer Institute?
Introduction - or: What is "MPEG Audio Layer-3"?
Today, efficient coding techniques are a must for cost-effective
processing of digital audio and video data by computers. Data
reduction of moving pictures and sound is a key technology for any
application with limited transmission or storage capacity. In the
recent years, a lot of progress has been achieved. While there (still)
exist several proprietary formats for audio and video coding, the
ISO/IEC standardisation body has released an international standard
("MPEG") for powerful audio and video coding tools (see: Overview
about the ISO-MPEG Standard - or: What is MPEG all about?).
Without data reduction, digital audio signals typically consist of 16
bit samples recorded at a sampling rate more than twice the actual
audio bandwidth (e.g. 44.1 kHz for Compact Disks). So you end up with
more than 1400 kbit to represent just one second of stereo music in CD
quality. By using MPEG audio coding, you may shrink down the original
sound data from a CD by a factor of 12, without losing sound quality.
Factors of 24 and even more still maintain a sound quality that is
significantly better than what you get by just reducing the sampling
rate and the resolution of your samples. Basically, this is realized
by "perceptual coding" techniques addressing the perception of sound
waves by the human ear (see: Basics of Perceptual Audio Coding - or:
What is the trick?).
Using MPEG audio, one may achieve a typical data reduction of 1:4
by Layer 1 (corresponds with 384 kbps for a stereo signal), 1:6...1:8
by Layer 2 (corresponds with 256..192 kbps for a stereo signal),
1:10...1:12 by Layer 3 (corresponds with 128..112 kbps for a stereo
still maintaining the original CD sound quality.
By exploiting stereo effects and by limiting the audio bandwidth, the
coding schemes may achieve an acceptable sound quality at even lower
bitrates. Layer-3 is the most powerful member of the MPEG audio coding
family. For a given sound quality level, it requires the lowest
bitrate - or for a given bitrate, it achieves the highest sound
quality (see: Advanced Features of Layer-3 - or: Why does Layer-3
perform so well?).
Some typical performance data of Layer-3 are:
sound quality bandwidth mode bitrate reduction ratio
------------------------ --------- ------ ----------------- -----
"telephone sound" 2.5 kHz mono 8 kbps (*) 96:1
"better than shortwave" 4.5 kHz mono 16 kbps 48:1
"better than AM radio" 7.5 kHz mono 32 kbps 24:1
"similar to FM radio" 11 kHz stereo 56..64 kbps 26..24:1
"near-CD" 15 kHz stereo 96 kbps 16:1
"CD" >15 kHz stereo 112..128kbps 14..12:1
*: Fraunhofer uses a non-ISO extension of Layer-3 for enhanced
performance ("MPEG 2.5")
All in all, Layer-3 is the key for numerous low-bitrate, high-quality
sound applications (see: Applications - or: Layer-3, what is it good
Applications - or: Layer-3, what is it good for?
A key technology like Layer-3 is useful for a pretty large spectrum of
applications - practically almost any system with a limited channel
capacity may benefit from it. The following chapters identify some
main areas and list some companies that are actively exploiting the
Layer-3 technology. For product-related information, please contact
these companies directly.
Music Links via ISDN
Digital telephone networks (ISDN = Integrated Services Digital
Network) offer reliable dial-up links with two 64 kbps data channels
per basic rate adapter; other regional networks (in North-America) use
56 kbps data links. Transmission fees are often rather similar or
identical to the traditional analog phone lines - those allow to
transmit up to 28.8 kbps (V.34 modem) or even 32 kbps ("V.34+").
Using Layer-3, a low-cost narrowband ISDN connection allows to
transmit CD-quality sound. Audio professionals, like broadcasting
stations and sound studios, benefit from the "music-by-phone"
application in various ways. They save money, as they only pay
transmission fees for the actual time of usage (not 24 h a day in case
of a leased phone line) and for a rather small data channel (one ISDN
phone connector for a stereo music link). Radio stations increase the
attractiveness of their programs, as reporters transmit high-quality
takes (e.g. an interview) or live news without annoying "telephone
sound". And new applications become possible, e.g. a "virtual studio",
where remote artists may play along some preproduced material, without
actually travelling to the studio.
* In 1992, Radio FFN, a private broadcasting station in
Niedersachsen, Germany, replaced its leased phone lines with ISDN
and Layer-3 codecs, to transmit 8 local programs 20 min per day to
the central broadcasting studio. This move saved them transmission
fees of more than 300.000 US$ per year.
* As one of the first real-world trials, all private radio stations
of Germany very successfully used Layer-3 codecs during the Winter
Olympic Games in Albertville (France) as reporter links between
the various sporting events and their central studio in Meribel.
* At the International Music Festival 92 in Bergen, Arne Nordheim
composed a piece of music, where an organ in the church of
Trondheim played along with the symphony orchestra in Bergen; the
sound of the organ was transmitted via ISDN and a Layer-3 codec.
Since 1992, various manufacturers are providing equipment ("codecs")
for professional audio applications: AVT, Broadcast Electronics, CCS,
Dialog 4, Telos.
Digital Satellite Broadcasting
Pioneered by WorldSpace, a worldwide satellite digital audio
broadcasting system is under construction. Its name is "WorldStar",
and it will use three geostationary orbit satellites called "AfriStar
1" (21 East), "CaribStar 1" (95 West), and "AsiaStar 1" (105 East),
with AfriStar 1 being launched in mid-1998. The other satellites will
follow until mid-1999. Each satellite is equipped with three downlink
spot beams that are pointed so as to cover populations that provide
the greatest radio listener base. Each downlink uses TDM (time
division multiplexing) to carry 96 prime rate channels (16 kbps each).
The prime rate channels are combined to carry broadcast channels
ranging from 16 kbps to 128 kbps; the broadcast channels are coded
using MPEG Layer-3. The prime rate channels may even be dynamically
allocated to meet the demands of the broadcast service (e.g. 4
channels combined for 1 hour to allow FM quality stereo (64 kbps) for
the transmission of a concert with classic music, followed by 1 hour
with 4 separate news channels (16 kbps) in 4 different native
WorldSpace is offering channels on its three satellites for lease to
international and national broadcasters. Channel reservation
agreements already have been signed with a number of major
broadcasters, including Voice of America, Radio Nederland, the Kenya
Broadcasting Corporation, the national broadcasting authority of
Ghana, the national broadcasting authority of Zimbabwe, New Sky Media
of Korea, and RCN of Columbia. Nearly 1 billion $ in private financing
has been raised to cover acquisition of the satellites and for most of
the operational costs through full system implementation in 1999.
France«s Alcatel Espace is the spacecraft prime contractor and
supplies the telecommunications payload.
The radio receivers will be designed for maximum convenience of use at
a minimum cost. Low cost receiver will use a small compact patch
antenna, will require practically no pointing, and will tune
automatically to selected channels. Higher end receivers are also
envisioned. In a press release from 5. June 96 (Montreux,
Switzerland), WorldSpace declared that it has awarded production
contracts for two million receiver chips; the contracts were issued to
SGS-Thomson and ITT Intermetall, authorizing each company for an
initial production of one million receiver chip-sets.
ITT Intermetall has already gained Layer-3 knowhow by using its
mask-programmed DSP technology to develop a single-chip Layer-3
decoder named "MAS 3503 C". This chip supports only MPEG-1 Layer-3.
The Internet is a world-wide packet-switched network of computers
linked together by various types of data communications systems.
Professional Internet providers usually access the network through
rather high bit-rate links (e.g., primary rate ISDN with 2 Mbps or ATM
with up to 2 Gbps). However, the average consumer uses low cost, low
bit-rate connections (e.g., basic rate ISDN with 64 kbps or phone line
modems with 28.8 or 14.4 kbps). The actual transmission rate depends
on the current user load and the infrastructure of the part of the
Internet in use. From a client«s point of view, it may unpredictably
vary between zero and the maximum bit-rate of its network modem, with
an average bit-rate somewhere in between.
Without audio coding, downloading uncompressed high-quality audio
files from a remote Internet server would result in unfavourably long
transmission times. For example, with an average transmission rate of
28.8 kbaud (optimistic guess), a single 3-min stereo track from a CD
(31.7 Mbyte) would require a download time of more than 2 hours.
Therefore, audio on the Internet calls for an audio coding scheme that
maintains sound quality as far as possible and allows real-time
decoding on a large number of computer platforms without special
add-on hardware. Layer-3 fits very well into this scenario - real-time
players (like WinPlay3) are available. Intranets present an
interesting special case, as they usually provide sufficient bitrate
to allow a number of real-time audio links. Furthermore, our
experiments indicate that using the http protocol, a real-time
connection with 56 (112) kbps is possible with one (two) ISDN phone
If content providers are willing to add audio data onto their Internet
servers, they have to consider carefully the copyright aspects of the
music industry (e.g., artists, producers, record companies). They must
not violate these rights by their actions! In the framework of a
European project called MODE (for "Music-on-Demand"), we developed a
flexible protection scheme called MMP (for "MultiMedia protection
protocol") that effectively addresses this issue. Furthermore, MMP
allows to distribute real-time players "virtually free".
Audio servers may be used plainly for promotional purposes. E.g.,
museums may increase the attractiveness of their WWW pages by adding
some sound files, or mail-order services may add sound excerpts to
their server to increase their CD sales numbers. Opticom, a spin-off
from Fraunhofer, offers system solutions for this type of application.
In spring 1996 (CeBit Hannover), they successfully demonstrated an
"audio-on-demand" application via T-Online together with the Deutsche
Telekom and a broadcasting station, the Suedwestfunk Baden-Baden.
Another music sales systems has been developed by Cerberus Sound &
Vision. The company uses a personalized real-time Layer-3 player and a
proprietary encryption scheme to sell sound files via the Internet on
a "per song" base. Music servers and mirror sites are currently
located in London, New York, Tokyo and Rio; Melbourne and Berlin will
"Audio-on-the-Internet" is currently a very popular topic. It does not
only comprise audio file transfers with download times as low as
possible, but also streaming audio applications, like "Internet
Radio". As Layer-3 offers a sound quality "better than shortwave" at a
bitrate of 16 kbps (and, with some modifications, may even be useful
at 8 kbps), various companies currently work on this Internet subject
- e.g., Opticom or Telos.
In a partnership with Apple, Telos introduced in September 96 the
Audioactive technology to support "Internet Radio" applications with a
live audio input processed by a Layer-3 NetCoder Hardware.
NEW ! In December 96, Microsoft announced to support MPEG Layer-3 as
part of their NetShow multimedia server technology.
As first multimedia authoring tools, "Director Multimedia Studio 2"
and "SoundEdit 16" (from Macromedia) use Layer-3 to generate
compressed sound files for the "shockwave" format.
Layer-3 encoders and decoders are not only available as studio
equipment, but also as ISA-bus PC boards from Dialog 4, along with
application software, or as low-cost (decoder only) PC boards from
NSM; recording and playback tools are also available from Proton Data,
along with a special decoder module (called "CenLay3") that allows to
playback Layer-3 files via the parallel printer port. Proton Data has
also developed a "cutting tool" that allows to manipulate audio data
at Layer-3 level.
In addition, a file-oriented Layer-3 encoder and decoder (called
"L3ENC" and "L3DEC") is available as shareware for various platforms.
Registration is processed by Opticom. Please note that even for
registered users, the use of the shareware is limited to "personal
Real-time Layer-3 players
"WinPlay3" allows the decoding simply by software on any Pentium PC in
real time. A 80486 class CPU with a built-in floating-point-unit will
also allow some limited operation. For the availability of supported
modes, please refer to the following performance matrix:
Pentium 486DX4-133 486DX2-66 486DX-50 486DX-33
------- ---------- --------- -------- --------
MPEG-1 stereo ok ok - - -
MPEG-1 downmix* ok ok ok - -
MPEG-1 mono ok ok ok ok -
MPEG-2 stereo ok ok ok ok -
MPEG-2 downmix ok ok ok ok ok
MPEG-2 mono ok ok ok ok ok
*downmix: the original stereo signal will be played back as a mono signal
"MPEG-1" = "MPEG-1 Layer-3", i.e. sample rates 32, 44.1 or 48 kHz
"MPEG-2" = "MPEG-2 Layer-3", i.e. sample rates 16, 22.05 or 24 kHz
On a Pentium-90, WinPlay3 consumes less than 30 % of the CPU power to
decode Layer-3 stereo @ 44.1 kHz, or around 5 % of the CPU power to
decode Layer-3 mono @ 16 kHz.
At least, a 8-bit stereo sound card is required. For full quality
audio, a 16-bit card is recommended. The card«s MCI driver should
support sampling frequencies from 8 kHz to 48 kHz.
A standard VGA graphics card is required.
As WinPlay3 buffers up to 4 seconds of sound data due to the
limitations of the Microsoft Windows multitasking architecture, around
1 MByte free physical memory must be available.
WinPlay3 runs with the following operating systems: Microsoft Windows
3.1/3.11 (in extended 386 mode), Windows 95 und Windows NT (long file
names not yet supported).
WinPlay3 supports file play back of *.mp3 files and direct play from
an URL via HTTP. WinPlay3 can simply be integrated as an helper
application in common browsers, for example Netscape or Mosaic.
WinPlay3 is available at
http://www.iis.fhg.de/departs/amm/layer3/winplay3/. The unregistered
player is limited to a reproduction time of 20 sec, i.e. it will
playback each plain Layer-3 file only for this time. If you want to
use your player without limitation, you have to register your player
As many applications require a player that is "free" for the user, the
latest versions of WinPlay3 (starting with version 2.0) also support
the new "MMP" ("MultiMedia protection protocol") format.
MMP is a very flexible data format that may support the following
* "unlocking" of the 20 sec playback time limitation
* "copyright protection" by applying encryption methods to (part of)
* "title associated data" (e.g. ISRC code, user data)
* "expiry date" to allow only a limited use
More detailed information is available at
In a typical "audio-on-demand" application, the content provider may
"on-the-fly" convert its plain Layer-3 data into MMP data, by using a
"MMP tagger" software (available at Opticom). The client may use its
unregistered player to playback these files without limitation - the
player is "virtually free". The client need not pay fees - this issue
now may be covered at the server side.
MPEG Layer 3 Player
For Mac OS users, a real-time player called "MPEG Layer 3 Player" with
a similar look and feel (and similar features) like "WinPlay3" will be
released very soon. This new player will (finally!) replace the much
simpler (and somewhat buggy) pre-version 0.99 beta that has been
available from http://www.iis.fhg.de/departs/amm/layer3/macplay3/.
Layer-3 Sound on CD-ROMs
CD-ROMs (and hard disks) have become most popular to store
"multimedia" data. Even with the advent of the new DVD standard,
memory capacity will remain a precious resource for many applications.
For uncompressed stereo signals from a CD, more than 10 MByte are
necessary to store one minute of music. Using Layer-3, less than 1
MByte is enough for the same playing time. And significantly less
memory is necessary, if some limitations in performance are
acceptable. As CD-ROM readers (and pretty soon, writers too) have
already gained a significant market share, typical applications focus
today on storing compressed sound files on CD-ROMs, introducing more
or better sound tracks into the product. Real application examples are
video games, music catalogues or encyclopedias with sound excerpts
(e.g., "MusicFinder" by Sygna), or talking books for blind people.
NEW !!! Since fall 96, Bertelsmann is selling their new CD-ROM
encyclopedia "Discovery 97" providing information to around 100.000
key words, with rich multimedia information (e.g. more than 2400
coloured photos and images, 41 interactive maps, more than 30 minutes
of movie clips, 27 slide shows) including 150 minutes of sound tracks
coded with MPEG Layer-3.
Layer-3 Sound on Silicon
Up to now, solid-state memories (RAMs, Flash-ROMs) are only used as
audio storage devices in special (niche) applications, as the costs
per byte are much higher than with other types of media
(magneto-optical disks or magnetic tapes). Speech announcement systems
for mass transit vehicles (e.g., busses, subways or trains) are an
example for such special applications, as the rough environment
requires to use ROM based memories. Since 1993, Meister Electronic
manufactures speech announcement systems with Layer-3, significantly
reducing the precious memory capacity and, at the same time,
significantly improving the sound quality (compared with their older
64 kbps PCM "phone sound").
Today, PC-Cards with Flash-ROMs are available, offering a memory
capacity up to 100 MByte and more, but at prohibitive high costs for a
consumer application. Here, further advances in memory and card
technology may trigger a new interesting market segment of
"audio-chip-card"-applications. At a press conference in August 95 in
Munich, Siemens Germany announced the advent of a new cost-effective
ROM technology called the "ROS chip" (ROS = Record-on-Silicon). The
first generation of ROS chips will be in production in 1997, with a
storage capacity of 64 Mbit; a next generation with 256 Mbit as well
as a one-time user programmable version will follow. The ROS chips
will be embedded in the new "MultiMedia-Card" from Siemens, a
cost-effective card media that will store data, text, graphics, images
and sound. Siemens has already demonstrated a battery-powered audio
player using a prototype "Audio-Card" containing sound tracks coded
General Questions and Answers
* Q: O.K., Layer-3 is obviously a key to many applications. Where
are its limitations?
* A: Well, Layer-3 is a perceptual audio coding scheme, exploiting
the properties of the human ear, and trying to maintain the
original sound quality as far as possible.
In contrast, a dedicated speech codec exploits the properties of
the human vocal tract, trying to maintain the intelligibility of
the voice signals as far as possible. Advanced speech coding
schemes (e.g., CS-ACELP [LD-CELP] as standardised by ITU as
G.723.1 [G.728]) achieve a useful voice reproduction at bitrates
as low as 5.3  kbps, with a codec delay below 40  ms. At
such very low bitrates, they behave superior to Layer-3 for pure
voice signals, and they offer the low delay that is necessary for
full- duplex voice communications.
In the framework of MPEG-4, scalable audio coding schemes are
devised that combine speech coding and perceptual audio coding.
* Q: You mentioned the codec delay. May I have some figures?
* A: Well, the standard gives some figures of the theoretical
Layer-1: 19 ms (<50 ms)
Layer-2: 35 ms (100 ms)
Layer-3: 59 ms (150 ms)
Practical values are significantly above that. As they depend on
the implementation, precise figures are hard to give. So the
numbers in brackets are just rough thumb values - real codecs may
show even higher values. So yes, there are certain applications
that may suffer from such a delay (like feedback links for remote
reporter units). For many other applications (like the ones
mentioned above), delay is of minor interest.
Overview about the ISO-MPEG Standard - or: What is MPEG all about?
* Q: What is "MPEG"?
* A: MPEG is the "Moving Picture Experts Group", working under the
joint direction of the International Standards Organization (ISO)
and the International Electro-Technical Commission (IEC). This
group works on standards for the coding of moving pictures and
audio. MPEG has created its own homepage, providing information on
the what, where, when and how of the standards.
* Q: What is MPEG-1, -2, and so on?
* A: MPEG approaches the growing need for multimedia standards
step-by-step. Today, three main "steps" are defined (MPEG-1,
+ MPEG-1: "Coding of Moving Pictures and Associated Audio for
Digital Storage Media at up to about 1.5 Mbit/s"
+ MPEG-2: "Generic Coding of Moving Pictures and Associated
+ MPEG-3: originally planned mainly for HDTV applications;
later on, it was merged into MPEG-2
+ MPEG-4: "Coding of Audio-Visual Objects"
* Q: Are MPEG-3 and Layer-3 the same thing?
* A: No! Layer-3 is a powerful audio coding scheme which certainly
is part of the MPEG standard. Layer-3 is defined within the audio
part of both existing international standards, MPEG-1 and MPEG-2.
So please do not mix audio layers and MPEG standards!
* Q: What is the status of MPEG-1?
* A: Work on MPEG-1 is finished. The first three parts are
standardized since 1992. MPEG-1 consists of five parts:
+ IS-11172-1 ("System") describes synchronization and
multiplexing of video and audio signals.
+ IS-11172-2 ("Video") describes compression of video signals,
focussing on progressive scan video (and mainly aiming at
+ IS-11172-3 ("Audio") describes a generic audio coding family,
with three hierarchically compatible members (called
"Layer-1", "Layer-2" and "Layer-3").
+ IS-11172-4 ("Compliance Testing") describes procedures for
determining the characteristics of coded bitstreams and the
decoding process and for testing compliance with the
requirements stated in the other parts.
+ DTR-11172-5 ("Software Simulation") is a technical report
about a full software implementation of the first three parts
* Q: What is the status of MPEG-2?
* A: MPEG-2 currently consists of nine parts. The first three parts
are standardized since 1994, with some amendments included later
on. Other parts are at different levels of completion.
+ IS-13818-1 ("System") describes synchronization and
multiplexing of video and audio signals; it is also
standardised by ITU-T as H.222.
+ IS-13818-2 ("Video") describes a generic video coding tool
set, supporting interlaced scan; it is also standardised by
ITU-T as H.262.
+ IS-13818-3 ("Audio") describes a backward compatible
extension of MPEG-1 for multichannel audio coding ("surround
sound", "multilingual sound") and a non-backward compatible
extension to lower sample rates, to support sound
applications with limited audio bandwidth requirements.
+ IS-13818-4 ("Conformance Testing") describes procedures for
determining the characteristics of coded bitstreams and the
decoding process and for testing compliance with the
requirements stated in the other parts.
+ DTR-13818-5 ("Software Simulation") is a technical report
about a full software implementation of the first three parts
+ IS-13818-6 ("System Extensions - Digital Storage Media
Command and Control (DSM-CC))" describes a set of protocols
for client-server applications
+ CD-13818-7 ("Audio, Non-Backwards-Compatible (NBC) - Coding")
describes an improved audio coding scheme for mono- and
stereophonic signals as well as for multichannel sound
+ 13818-8 ("Video, extension to 10-bit input samples") has been
withdrawn, due to insufficient interest.
+ IS-13818-9 ("Real-Time Interface Specification for Low-Jitter
Applications") defines timing constraints on the real-time
delivery of MPEG-2 transport bitstreams.
+ WD-13818-10 ("Conformance Extensions - DSM-CC") describes the
addendum to IS 13818-4 for DSM-CC
* Q: "NBC audio"?" What is the motivation for this working group?
What are the results?
* A: Well, during the work for multichannel audio coding
(IS-13818-3), it turned out that backwards compatible (BC) schemes
suffer from the matrixing process. Matrixing is required to allow
a MPEG-1 decoder to playback all surround channels via its two
stereophonic channels. Unfortunately, some of the introduced
quantisation noise may become audible after dematrixing. All in
all, during an ISO listening test in spring 1994, BC multichannel
coding performed poorer, compared to non-ISO coding schemes (e.g.,
Dolby«s AC-3). So the NBC working group currently develops a new
audio coding scheme. NBC audio achieves a significant better
performance, not only for multichannel surround sound, but even
for monophonic signals (here targeting "true transparency" at 64
kbps). In spring 1996, ISO performed a listening test for
5-channel surround sound, and NBC audio using a total bit-rate of
320 kbps scored better than Layer-2 BC at a bit-rate of 640 kbps.
NBC audio will also become one of the MPEG-4 audio coding
* Q: How do I get the MPEG documents?
* A: Well, you may contact ISO, or you order it from your national
standards body. E.g., in Germany, please contact DIN.
* Q: Is some public C source available?
* A: Well, there is "public C source" available on various sites,
e.g. at ftp://ftp.fhg.de/pub/layer3/ or at
/ . This code has been written mainly for explanation purposes, so
do not expect too much performance.
Some Basics about MPEG Audio - or: What about Layer-1, Layer-2, Layer-3?
* Q: Talking about MPEG audio, I always hear "Layer 1, 2 and 3".
What does it mean?
* A: MPEG describes the compression of audio signals using high
performance perceptual coding schemes. It specifies a family of
three audio coding schemes, simply called Layer-1, Layer-2, and
Layer-3. From Layer-1 to Layer-3, encoder complexity and
performance (sound quality per bitrate) are increasing.
The three codecs are compatible in a hierarchical way, i.e. a
Layer-N decoder may be able to decode bitstream data encoded in
Layer-N and all Layers below N (e.g., a Layer-3 decoder may accept
Layer-1,-2,-3, whereas a Layer-2 decoder may accept only Layer-1
* Q: So we have a family of three audio coding schemes. What does
the MPEG standard define, exactly?
* A: For each Layer, the standard specifies the bitstream format and
the decoder. To allow for future improvements, it does not specify
the encoder, but an informative chapter gives an example for an
encoder for each Layer.
* Q: What have the three audio Layers in common?
* A: All Layers use the same basic structure. The coding scheme can
be described as "perceptual noise shaping" or "perceptual subband
/ transform coding". The encoder analyzes the spectral components
of the audio signal by calculating a filterbank (transform) and
applies a psychoacoustic model to estimate the just noticeable
noise-level. In its quantization and coding stage, the encoder
tries to allocate the available number of data bits in a way to
meet both the bitrate and masking requirements.
The decoder is much less complex. Its only task is to synthesize
an audio signal out of the coded spectral components.
All Layers use the same analysis filterbank (polyphase with 32
subbands). Layer-3 adds a MDCT transform to increase the frequency
All Layers use the same "header information" in their bitstream,
to support the hierarchical structure of the standard.
All Layers have a similar sensitivity to biterrors. They use a
bitstream structure that contains parts that are more sensitive to
biterrors ("header", "bit allocation", "scalefactors", "side
information") and parts that are less sensitive ("data of spectral
All Layers support the insertion of programm-associated
information ("ancillary data") into their audio data bitstream.
All Layers may use 32, 44.1 or 48 kHz sampling frequency.
All Layers are allowed to work with similar bitrates:
Layer-1: from 32 kbps to 448 kbps
Layer-2: from 32 kbps to 384 kbps
Layer-3: from 32 kbps to 320 kbps
The last two statements refer to MPEG-1; with MPEG-2, there is an
extension for the sampling frequencies and bitrates (see below).
* Q: What are the main differences between the three Layers, from a
* A: From Layer-1 to Layer-3, complexity increases (mainly true for
the encoder), overall codec delay increases, and performance
increases (sound quality per bitrate).
* Q: What are the main differences between MPEG-1 and MPEG-2 in the
* A: MPEG-1 and MPEG-2 use the same family of audio codecs, Layer-1,
-2 and -3. The new audio features of MPEG-2 are a "low sample rate
extension" to address very low bitrate applications with limited
bandwidth requirements (the new sampling frequencies are 16, 22.05
or 24 kHz, the bitrates extend down to 8 kbps), and a
"multichannel extension" to address surround sound applications
with up to 5 main audio channels (left, center, right, left
surround, right surround) and optionally 1 extra "low frequency
enhancement (LFE)" channel for subwoofer signals; in addition, a
"multilingual extension" allows the inclusion of up to 7 more
* Q: Is this all compatible to each other?
* A: Well, more or less, yes - with the execption of the low sample
rate extension. Obviously, a pure MPEG-1 decoder is not able to
handle the new "half" sample rates.
* Q: You mean: compatible!? With all these extra audio channels?
* A: Compatibility has been a major topic during the MPEG-2
definition phase. The main idea is to use the same basic bitstream
format as defined in MPEG-1, with the main data field carrying two
audio signals (called L0 and R0) as before, and the ancillary data
field carrying the multichannel extension information. Without
going further into details, two terms should be explained here:
"forwards compatible": the MPEG-2 decoder has to accept any MPEG-1
audio bitstream (that represents one or two audio channels)
"backwards compatible": the MPEG-1 decoder should be able to
decode the audio signals in the main data field (L0 and R0) of the
MPEG-2 bitstream "Matrixing" may be used to get the surround
information into L0 and R0: L0 = left signal + a * center signal +
b * left surround signal R0 = right signal + a * center signal + b
* right surround signal Therefore, a MPEG-1 decoder can reproduce
a comprehensive downmix of the full 5- channel information. A
MPEG-2 decoder uses the multichannel extension information (3 more
audio signals) to reconstruct the five surround channels.
* Q: In your footnotes, you indicate the use of some "non-ISO"
extension inside your Fraunhofer codec, called "MPEG 2.5", to
further improve the performance at very low bitrates (e.g. 8 kbps
mono). What do you mean by this?
* A: Oh, yes. Well, the MPEG-2 standard allows bitrates as low as 8
kbps, for the low sample rate extension. At such a low bitrate,
the useful audio bandwidth has to be limited anyway, e.g. to 3
kHz. Therefore, the actual sample rate could be reduced, e.g. to 8
kHz. The lower the sample rate, the better the frequency
resolution, the worse the time resolution, and the better the
ratio between control information and audio payload inside the
bitstream format. As the MPEG-2 standard defines 16 kHz as lowest
sample rate, we introduced a further extension, again dividing the
low sample rates of MPEG-2 by 2, i.e. we introduced 8, 11.025, and
12 kHz - and we named this extension to the extension "MPEG 2.5".
"Layer-3" performs significantly better with 8 kbps @ 8 kHz or 16
kbps @ 11 kHz than with 8 or 16 kbps @ 16 kHz.
Advanced Features of Layer-3 - or: Why does Layer-3 perform so well?
* Q: Well, I read your statement about "CD-like" performance,
achieved at a data reduction of 4:1 (or 384 kbps total bitrate)
with Layer-1, 6..8:1 (or 256..192 kbps total bitrate) with
Layer-2, and 12..14:1 (or 128..112 kbps total bitrate) with
Layer-3. Can you explain a little further?
* A: Well, each audio Layer extends the features of the Layer with
the lower number. The simplest form is Layer-1. It has been
designed mainly for the DCC (Digital Compact Cassette), where it
is used at 384 kbps (called "PASC"). Layer-2 has been designed as
a trade-off between complexity and performance. It achieves a good
sound quality at bitrates down to 192 kbps. Below, sound quality
suffers. Layer-3 has been designed for low bitrates right from the
start. It adds a number of "advanced features" to Layer-2: the
frequency resolution is 18 times higher, which allows a Layer-3
encoder to adapt the quantisation noise much better to the masking
threshold only Layer-3 uses entropy coding (like MPEG video) to
further reduce redundancy only Layer-3 uses a bit reservoir (like
MPEG video) to suppress artefacts in critical moments and Layer-3
may use more advanced joint-stereo coding methods
* Q: I see. Sounds to me as if Layer-3 is something like a
"Layer-2++". Now, tell me more about sound quality. How do you
* A: Today, there is no alternative to expensive listening tests.
During the ISO-MPEG process, a number of international listening
tests have been performed, with a lot of trained listeners. All
these tests used the "triple stimulus, hidden reference" method
and the "CCIR impairment scale" to assess the sound quality. The
listening sequence is "ABC", with A = original, BC = pair of
original / coded signal with random sequence, and the listener has
to evaluate both B and C with a number between 1.0 and 5.0. The
meaning of these values is: 5.0 = transparent (this should be the
original signal) 4.0 = perceptible, but not annoying (first
differences noticable) 3.0 = slightly annoying 2.0 = annoying 1.0
= very annoying
* Q: Listening tests are certainly an expensive task. Is there
really no alternative?
* A: Well, at least not today. Tomorrow may be different. To assess
sound quality with perceptual codecs, all traditional "quality"
parameters (like signal-to-noise ratio, total harmonic distortion,
bandwidth) are rather useless, as any codec may introduce noise
and distortions as long as these do not affect the perceived sound
quality. So, listening tests are necessary, and, if carefully
prepared and performed, they lead to rather reliable results.
Nevertheless, Fraunhofer-IIS works on the development and
standardisation of objective sound quality assessment tools, too.
And there is already a first product available (contact Opticom),
a real-time measurement tool that nicely supports the analysis of
perceptual audio codecs. If you need more information about the
Noise- to-Mask-Ratio (NMR) technology, feel free to contact
* Q: O.K., back to these listening tests and the performance
evaluation. Come on, tell me some results.
* A: Well, for more details you should study one of these AES papers
or the MPEG documents. For Layer-3, the main result is that it
always performed superior at low bitrates (64 kbps per audio
channel or below). Well, this is not completely surprising, as
Layer-3 uses the same tool set as Layer-2, but with some
additional advanced coding features that all address the demands
of very low bitrate coding. One impressive example is the ISO-MPEG
listening test carried out in September 94 at NTT Japan (doc.
ISO/IEC JTC1/SC29/WG11 N0848, 11.Nov. 94). Another interesting
result is the conclusion of the task group TG 10/2 within the ITU-
R, which recommends the use of low bit-rate audio coding schemes
for digital sound-broadcasting applications (ITU-R doc. BS.1115).
* Q: Very interesting! Tell me more about this recommendation!
* A: The task group TG 10/2 finished its work in 10/93. The
recommendation defines three fields of broadcast applications and
recommends Layer-2 with 180 kbps per channel for distribution and
contribution links (20 kHz bandwidth, no audible impairments with
up to 5 cascaded codec), Layer-2 with 128 kbps per channel for
emission (20 kHz bandwidth), and Layer-3 with 60 (120) kbps for
mono (stereo) signals for commentary links (15 kHz bandwidth).
Basics of Perceptual Audio Coding - or: What is the trick?
Sorry - under construction...
References - or: Where to find more information?
For around 10 years, perceptual audio coding is a permanent topic at
various scientific conferences; e.g., the AES (Audio Engineering
Society) organizes two conventions per year. You may find the
following papers helpful:
1. Brandenburg, Stoll, et al.: "The ISO/MPEG-Audio Codec: A Generic
Standard for Coding of High Quality Digital Audio", 92nd AES,
Vienna Mar. 92, pp. 3336; revised version ("ISO-MPEG-1 Audio: A
Generic Standard...") published in the Journal of AES, Vol.42, No.
10, Oct. 94
2. Eberlein, Popp, et al.: "Layer-3, a Flexible Coding Standard",
94th AES, Berlin Mar. 93, pp. 3493 3) Church, Grill, et al.: "ISDN
and ISO/MPEG Layer-3 Audio Coding: Powerful New tools for
Broadcast and Audio Production", 95th AES, New York Oct. 93, pp.
3. Grill, Herre, et al.: "Improved MPEG-2 Audio Multi-Channel
Encoding", 96th AES, Amsterdam Feb. 94, pp. 3865
4. Witte, Dietz, et al.: "Single Chip Implementation of an ISO/MPEG
Layer-3 Decoder", 96th AES, Amsterdam Feb. 94, pp. 3805
5. Herre, Brandenburg, et al.: "Second Generation ISO/MPEG Audio
Layer-3 Coding", 98th AES, Paris Feb. 95
6. Dietz, Popp, et al.: "Audio Compression for Network Transmission",
99th AES, New York Oct. 95, pp. 4129
7. Brandenburg, Bosi: "Overview of MPEG-Audio: Current and Future
Standards for Low Bit-Rate Audio Coding, 99th AES, New York Oct.
95, pp. 4130
8. Buchta, Meltzer, et al.: "The WorldStar Sound Format", 101st AES,
Los Angeles Nov. 96, pp. 4385
9. Bosi, Brandenburg, et al: "ISO/IEC MPEG-2 Advanced Audio Coding",
101st AES, Los Angeles Nov. 96, pp. 4382
Please note that these papers are not available electronically. You
have to order the preprints ("pp. xxxx") directly from the AES.
* AES, 60 East 42nd Street, Suite 2520 New York, NY 10165-2520, USA
fax: +1 212 682 0477
* AVT Audio Video Technologies GmbH, Rathsbergstra§e 17
D-90411 Nrnberg, Germany
fax: +49 911 5271 100
contact: Wolfgang Peters
* Bertelsmann Publishing, Neumarkter Stra§e 18
D-81664 Mnchen, Germany
fax: +49 89 43189 737
* Broadcast Electronics Inc, 4100 N 24th St.
Quincy, IL 62305-3606, USA
fax: +1 217 224 9607
* CCS Corporate Computer Systems Europe GmbH, Ludwigstr. 45
D-85396 Hallbergmoos, Germany
fax: +49 811 55 16 55
* Cerberus Central Ltd, 84 Marylebone High Street
London W1M 3DE, UK
fax: +44 171 637 3842
* Deutsche Telekom AG, Technologiezentrum Darmstadt
Aussenstelle Berlin, Abteilung EK 21
Oranienburger Str. 70, D-10117 Berlin, Germany
fax: +49 30 2845 4146
* Dialog 4 System Engineering GmbH, Monreposstr. 55
D-71634 Ludwigsburg, Germany
fax: +49 7141 22667
* DIN Beuth Verlag, Auslandsnormen
D-10772 Berlin, Germany
fax: +49 30 2601 1231
* Fraunhofer-IIS, Am Weichselgarten 3
D-91058 Erlangen, Germany
contact: Harald Popp
fax: +49 9131 776 399
* ISO Central Secretariat, Case postale 56,
CH-1211 Geneva 20, Switzerland
fax: +41 22 733 3430
* ITT Intermetall GmbH, Hans-Bunte-Str. 19
D-79108 Freiburg, Germany
fax: +49 761 517 2395
* Macromedia Inc., 600 Townsend
San Francisco, CA 94103, USA
fax: +1 415 626 0554
* Meister Electronic GmbH, Klner Str. 37
D-51149 Kln, Germany
fax: +49 2203 1701 30
* Microsoft Inc., One Microsoft Way
Redmond, WA 98052 - 6399
* NSM, Im Tiergarten 20 - 30
D-55411 Bingen am Rhein, Germany
contact: Mr. Ballhorn
fax: +49 6721 407 519
* Opticom, Am Weichselgarten 7
D-91058 Erlangen, Germany
fax: +49 9131 691325
* Proton Data, Marrensdamm 12 b
D-24944 Flensburg, Germany
fax: +49 461 3816948
* Siemens AG Halbleiter, P.O. Box 80 17 09
D-81617 Muenchen, Germany
fax: +49 89 4144 4697
* Sygna A/S, P.O.Box 191
N-5801 Sogndal, Norway
fax: +47 5767 6190
* Telos Systems, 2101 Superior Avenue
Cleveland, OH 44114, USA
fax: +1 216 241 4103
* WorldSpace 11 Dupont Circle, N.W., 9th Floor
Washington, DC 20036, USA
fax: +1 202 884 7900
About us - or: What is going on at our Fraunhofer Institute?
* Q: Who is or was Fraunhofer? And what does your institute do?
* A: As researcher, inventor and entrepreneur, Joseph von Fraunhofer
(1787 - 1826) won high acclaim for his scientific and commercial
achievements. When the Fraunhofer-Gesellschaft was founded in
Munich in 1949, his name was chosen as the "guiding light" of the
Today, the Fraunhofer-Gesellschaft employs a staff of around 8.000
persons and operates 46 research institutes in Germany and one
resource centre in the United States, with a research volume of
around 1 billion DM. 70 % of its income is obtained by contract
research for public authorities as well as for industrial clients.
The Fraunhofer Institut Integrierte Schaltungen (IIS) was founded
in Erlangen in 1985. It is headed by Prof. Dr.Ing. Dieter Seitzer
and Dr. Heinz Gerhuser. Today, a staff of 160 persons works on
projects in the field of information electronics, developing
microelectronic solutions at chip-, board- and system level. In
its department "Audio & Multimedia", headed by Dr. Karlheinz
Brandenburg, around 40 engineers concentrate on the development
and real-time implementation of signal processing algorithms in
the field of audiovisual communications.
* Q: So you focus on "contract research". What does this mean
* A: Simply put: we have to earn our money. In case of our
institute, we are funded by public money for less than 20 % - the
rest of our budget has to be financed by research & development
projects. You may call this work "applied research", i.e. in
contrast to a university, we focus on real-world applications, and
in contrast to an engineeering office, we focus on
state-of-the-art applications that bear some technical risks (and
therefore need some further research). With other words, we are
always trying to stay at the leading edge of technology. Take
audio coding as an example. We started in 1987, in a close
cooperation with the University of Erlangen, to develop an
advanced audio coding scheme for future broadcast services (Eureka
147, DAB radio). In 1991, our algorithm ("Layer-3") became the
most powerful member of audio coding schemes of the international
ISO-MPEG standard. Since then, we work on industrial applications
as well as on further audiovisual research projects, e.g. MPEG-4
scalable audio coding, MPEG-2 NBC audio coding, or MPEG-4
* Q: I am interested in your Layer-3 technology. What can you do for
* A: Well - basically, you may use our knowhow as a cost-effective
road to your application. We expect a certain renumeration for our
development work that we carried out in advance. We call this a
"know-how share". In addition, you may want us to work on some
special R&D tasks for you, so you have to pay for this extra
effort, too. This is the principle. In case of Layer-3, we have
advanced simulation sources (C) for encoder and decoder as well as
DSP source and assembler code for decoders on DSP 5600x
(Motorola), DSP 32C (AT&T), TMS320C30 (TI), and MAS 3503 C (ITT),
and for encoders on a hybrid solution (32C + 5600x) as well as on
a pure 5600x (2 DSPs) solution. We expect a single 5630x Layer-3
encoder until the end of 1996. Depending on your specific
technical needs, the knowhow-share sum may range from several
10.000.- $ to more than 100.000.- $. In any case, we expect
significantly more money for the encoder, as this is the part that
is responsible for the performance of a Layer-3 system (and so it
is the part where most of our knowhow is concentrated). So you
know the framework. We are open for any discussion and any new
ideas - so feel free to contact us.
Oh - by the way you are interested in some rough ASIC estimations
for a Layer-3 stereo decoder. You will need a computation power of
around 12 MIPs, a Data ROM of around 2.5 Kwords, a Data RAM of
around 4.5 Kwords, and a Programm ROM of around 2 to 4 Kwords
(depending on the instruction set). The word length should be 20
bit, at least.
* Q: What else do I have to keep in mind, if I want to use Layer-3
in my application? Are there patents involved? How may I address
* A: You are right. For all MPEG audio coding schemes, patent rights
exist. Using MPEG audio, you use these rights - and in order not
to violate them, you should establish a license contract with the
patent holders. This is true for all MPEG audio Layers. In case of
Layer-3, there are currently two entities that may give licenses,
Thomson Multimedia, Paris, and Fraunhofer-IIS, Erlangen. Due to an
agreement between them, Thomson is in charge of consumer-oriented
applications, and Fraunhofer-IIS is in charge of
professional-oriented applications. License contracts typically
address only the patent issue. Due to the rules of ISO-MPEG, the
license has to be given non-exclusively on fair and reasonable
terms. Of course, details depend on the specific business model.
So there are four steps for a Layer-3 application. First, defining
the technical requirements and finding the most cost-effective
road to meet them. Second, following that road to the final
solution. Third, defining the license rules depending on the
business model. Four, signing the resulting license contract.
Fraunhofer Institut Integrierte Schaltungen IIS, Am Weichselgarten 3,
D-91058 Erlangen, Germany, Fax: +49-9131-776-399
FAQ, 29. Februar 1997, by Harald Popp
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