
  Frequently Asked Questions about MPEG Audio Layer-3
   

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, 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. 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.  
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 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 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.  
  
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 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 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/laye
r3/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.  
  
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., 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 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/aud
io/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:   
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  
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. 3743  
Grill, Herre, et al.: "Improved MPEG-2 Audio Multi-Channel Encoding",
96th AES, Amsterdam Feb. 94, pp. 3865  
Witte, Dietz, et al.: "Single Chip Implementation of an ISO/MPEG
Layer-3 Decoder", 96th AES, Amsterdam Feb. 94, pp. 3805  
Herre, Brandenburg, et al.: "Second Generation ISO/MPEG Audio Layer-3
Coding", 98th AES, Paris Feb. 95  
Dietz, Popp, et al.: "Audio Compression for Network Transmission",
99th AES, New York Oct. 95, pp. 4129  
Brandenburg, Bosi: "Overview of MPEG-Audio: Current and Future Standards
for Low Bit-Rate Audio Coding, 99th AES, New York Oct. 95, pp. 4130

Buchta, Meltzer, et al.: "The WorldStar Sound Format", 101st AES,
Los Angeles Nov. 96, pp. 4385  
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, Rathsbergstrae 17 
D-90411 Nrnberg, Germany
fax: +49 911 5271 100
contact: Wolfgang Peters
email: WPeters@avt-nbg.de
http://www.avt-nbg.de

Bertelsmann Publishing, Neumarkter Strae 18
D-81664 Mnchen, 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, Klner Str. 37
D-51149 Kln, 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 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 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   

