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Frequently Asked Questions about MPEG Audio Layer-3, Fraunhofer-IIS, and all the rest...

Version 2.83

This text will be continously upgraded: step by step, more answers and more information will be included. Yes, we
definitely know that there are a lot more questions to answer! But we cannot do that all at once. So, some parts may
remain "under construction" for a while, and other parts may be modified due to new results of our research work or new
applications. You find the latest release at

http://www.iis.fhg.de/departs/amm/layer3/sw/ or
ftp://ftp.fhg.de/pub/layer3/l3faq.html

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, Layer-3?
* Advanced Features of Layer-3 - or: Why does Layer-3 perform so well?
* 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 signal),

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 stero 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 for?).

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.

Examples:
* 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 tongues).

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. Franceґs 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.

Audio-on-Demand

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 clientґs 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 line(s).

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 follow soon.

"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 edition" purposes.

Real-time Layer-3 players

WinPlay3

"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 cardґs 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 with Opticom.

MMP

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 functions:
* "unlocking" of the 20 sec playback time limitation
* "copyright protection" by applying encryption methods to (part of) the data
* "title associated data" (e.g. ISRC code, user data)
* "expiry date" to allow only a limited use

More detailed information is available at http://www.iis.fhg.de/departs/amm/layer3/mmp/.

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 with MPEG-Layer-3.

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 [16] kbps, with
a codec delay below 40 [1] 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 minimum delay:
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-2, MPEG-4).

* 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 Audio Information"
* 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
"Video-on-CD" applications).
* 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 of MPEG-1.

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 of MPEG-2.
* 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., Dolbyґs 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 algorithms.

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
ftp://ftp.tnt.uni-hannover.de/pub/MPEG/audio/mpeg2/public_software/ . 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 and -2.)
* 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 resolution.
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 components").
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 global view?
* 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 audio part?
* 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 audio channels.
* 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? Please explain!
* 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 assess that?
* 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 nmr@iis.fhg.de.
* 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)
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. 3743
4. Grill, Herre, et al.: "Improved MPEG-2 Audio Multi-Channel Encoding", 96th AES, Amsterdam Feb. 94, pp. 3865
5. Witte, Dietz, et al.: "Single Chip Implementation of an ISO/MPEG Layer-3 Decoder", 96th AES, Amsterdam Feb. 94, pp.
3805
6. Herre, Brandenburg, et al.: "Second Generation ISO/MPEG Audio Layer-3 Coding", 98th AES, Paris Feb. 95
7. Dietz, Popp, et al.: "Audio Compression for Network Transmission", 99th AES, New York Oct. 95, pp. 4129
8. Brandenburg, Bosi: "Overview of MPEG-Audio: Current and Future Standards for Low Bit-Rate Audio Coding, 99th AES, New
York Oct. 95, pp. 4130
9. Buchta, Meltzer, et al.: "The WorldStar Sound Format", 101st AES, Los Angeles Nov. 96, pp. 4385
10. 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.

Addressess

* AES, 60 East 42nd Street, Suite 2520 New York, NY 10165-2520, USA
fax: +1 212 682 0477
email: hq@aes.org
http://www.aes.org/
* AudioActive
http://www.audioactive.com/
* AVT Audio Video Technologies GmbH, RathsbergstraЯe 17
D-90411 Nьrnberg, Germany
fax: +49 911 5271 100
contact: Wolfgang Peters
email: WPeters@avt-nbg.de
http://www.avt-nbg.de
* Bertelsmann Publishing, Neumarkter StraЯe 18
D-81664 Mьnchen, Germany
fax: +49 89 43189 737
email: 72662.3126@compuserve.com
http://www.bep.de/
* Broadcast Electronics Inc, 4100 N 24th St.
Quincy, IL 62305-3606, USA
fax: +1 217 224 9607
email: bdcast@bdcast.com
http://www.marti.bdcast.com/
* CCS Corporate Computer Systems Europe GmbH, Ludwigstr. 45
D-85396 Hallbergmoos, Germany
fax: +49 811 55 16 55
email: info@ccs-europe.com
http://www.ccs-europe.com/
* Cerberus Central Ltd, 84 Marylebone High Street
London W1M 3DE, UK
fax: +44 171 637 3842
email: mail@cdj.co.uk
http://www.cdj.co.uk/
* 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
email: info@dialog4.com
http://www.dialog4.com/
* DIN Beuth Verlag, Auslandsnormen
D-10772 Berlin, Germany
fax: +49 30 2601 1231
email: postmaster@din.de
* Fraunhofer-IIS, Am Weichselgarten 3
D-91058 Erlangen, Germany
contact: Harald Popp
fax: +49 9131 776 399
email: layer3@iis.fhg.de
http://www.iis.fhg.de/departs/amm/layer3/
* ISO Central Secretariat, Case postale 56,
CH-1211 Geneva 20, Switzerland
fax: +41 22 733 3430
email: central@isocs.iso.ch
http://www.iso.ch/
* ITT Intermetall GmbH, Hans-Bunte-Str. 19
D-79108 Freiburg, Germany
fax: +49 761 517 2395
email: info@itt-sc.de
* Macromedia Inc., 600 Townsend
San Francisco, CA 94103, USA
fax: +1 415 626 0554
http://www.macromedia.com/
* Meister Electronic GmbH, Kцlner Str. 37
D-51149 Kцln, Germany
fax: +49 2203 1701 30
* Microsoft Inc., One Microsoft Way
Redmond, WA 98052 - 6399
http://www.microsoft.com/corpinfo/PRESS/1996/Dec96/ntshw2pr.htm
* MODE
http://www.mode.net/
* MPEG
http://www.cselt.stet.it/mpeg/
* NSM, Im Tiergarten 20 - 30
D-55411 Bingen am Rhein, Germany
contact: Mr. Ballhorn
fax: +49 6721 407 519
http://www.nsm.de/nsm_it/
* Opticom, Am Weichselgarten 7
D-91058 Erlangen, Germany
fax: +49 9131 691325
email: info@opticom.de
http://www.opticom.de
* Proton Data, Marrensdamm 12 b
D-24944 Flensburg, Germany
fax: +49 461 3816948
email: proton.data@t-online.de
* Siemens AG Halbleiter, P.O. Box 80 17 09
D-81617 Muenchen, Germany
fax: +49 89 4144 4697
email: Christine.Born@hl.siemens.de
* Sygna A/S, P.O.Box 191
N-5801 Sogndal, Norway
fax: +47 5767 6190
email: bach@sygna.no
http://www.mode.net/partners/sygna.html
* Telos Systems, 2101 Superior Avenue
Cleveland, OH 44114, USA
fax: +1 216 241 4103
email: info@zephyr.com
http://www.zephyr.com/
* WorldSpace 11 Dupont Circle, N.W., 9th Floor
Washington, DC 20036, USA
fax: +1 202 884 7900
email: gene@mail.worldspace.com
http://www.worldspace.com

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 association.
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 Gerhдuser. 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 exactly?
* 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 audiovisual terminals.
* Q: I am interested in your Layer-3 technology. What can you do for me?
* 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 this topic?
* 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, 19. December 1996, by Harald Popp

PS. Зачем же стену молотком долбить...