MPEG-FAQ 4.1: What is the Audio Layer 3 then ?
MPEG-FAQ 4.1:
What is the Audio Layer 3 then ?
Informations about MPEG Audio Layer-3
Version 1.51 - 1. 95
This text is organized as a kind of Mini-FAQ (Frequently Asked
Questions). It covers several topics:
1. ISO-MPEG Standard
2. MPEG Audio Codec Family ("Layer 1, 2, 3")
3. Applications
4. Products
5. Support by Fraunhofer-IIS
6. Shareware Information
For further comments and questions regarding Layer-3, please contact:
- layer3@iis.fhg.de
For further informations about MPEG, you may also like to contact:
- phade@powerweb.de
1. ISO-MPEG Standard
Q: What is MPEG, exactly?
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 associated audio.
Q: What is the status of MPEG's work, then? What about MPEG-1, -2, and so
on?
A: MPEG approaches the growing need for multimedia standards step-by-
step. Today, three "phases" are defined:
MPEG-1:"Coding of Moving Pictures and Associated Audio for
Digital Storage Media at up to about 1.5 MBit/s"
Status: International Standard IS-11172, completed in 10.92
MPEG-2:"Generic Coding of Moving Pictures and Associated
Audio"
Status: International Standard IS-13818, completed in 11.94
MPEG-3: does no longer exist (has been merged into MPEG-2)
MPEG-4: "Very Low Bitrate Audio-Visual Coding"
Status: Call for Proposals first deadline 1. 10. 95
Q: MPEG-1 and MPEG-2 are ready-for-use. How do the standards look like?
A: Both standards consist of 4 main parts.
The structure is the same for MPEG-1 and MPEG-2.
-1: System describes synchronization and multiplexing of video and audio
-2: Video describes compression of video signals
-3: Audio describes compression of audio signals
-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.
Q: How do I get the MPEG documents?
A: You order it from your national standards body.
E.g., in Germany, please contact:
DIN-Beuth Verlag, Auslandsnormen
Mrs. Niehoff, Burggrafenstr. 6, D-10772 Berlin, Germany
Phone: +49-30-2601-2757, Fax: +49-30-2601-1231
2. MPEG Audio Codec Family ("Layer 1, 2, 3")
Q: Talking about MPEG audio coding, I heard a lot about "Layer 1, 2 and 3".
What does it mean, exactly?
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,-2,-3, with increasing encoder complexity
and performance (sound quality per bitrate) from 1 to 3.
The three codecs are compatible in a hierarchical way, i.e. a Layer-N
decoder is 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 and -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 or 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:
"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),
"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: A lot of new stuff! 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, three terms can
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: I heard something about a new NBC mode for MPEG-2 audio? What does
it mean?
A: "NBC" stands for "non-backwards compatible". During the development
of the backwards compatible MPEG-2 standard, the experts encountered
some trouble with the compatibility matrix. The introduced quantisation
noise may become audible after dematrixing. Although some clever
strategies have been devised to overcome this problem, the question
remained how much better a non-compatible multichannel codec might
perform.
So ISO-MPEG decided to address that issue in a "NBC" working group -
among the proponents are AT&T, Dolby, Fraunhofer, IRT, Philips, and
Sony. Their work will lead to an addendum to the MPEG-2 standard
(13818-8).
Q: O.K., that should do for a first overview. Are there some papers for a more
detailed information?
A: Sure! You'll find more technical informations about MPEG audio coding
in a variety of AES papers (AES = Audio Engineering Society). The AES
organizes two conventions per year, and perceptual audio coding has been
a topic since the middle of the 80s. Some interesting papers might be:
K. Brandenburg, G. 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
S. Church, B. 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
E. Eberlein, H. Popp, et al.: "Layer-3, a Flexible Coding Standard",
94th AES, Berlin Mar. 93, pp. 3493
B. Grill, J. Herre, et al.: "Improved MPEG-2 Audio Multi-Channel
Encoding", 96th AES, Amsterdam Feb. 94, pp. 3865
J. Herre, K. Brandenburg, et al.: "Second Generation ISO/MPEG
Audio Layer-3 Coding", 98th AES, Paris Feb. 95
F.-O. Witte, M. Dietz, et al.: "'Single Chip Implementation of an
ISO/MPEG Layer-3 Decoder", 96th AES, Amsterdam Feb. 94, pp.
3805
For ordering informations, contact:
AES
60 East 42nd Street, Suite 2520
New York, NY 10165-2520, USA
phone: (212) 661-8528, fax: (212) 682-0477
Another interesting publication: the "Proceedings of the Sixth Tirrenia
International Workshop on Digital Communications", Tirrenia Sep. 93,
Elsevier Science B.V. Amsterdam 94 (ISBN 0 444 81580 5).
An excellent tutorial about MPEG-2 has recently been published in a
German technical journal (Fernseh- und Kino-Technik); part 4, by E. F.
Schroeder and J. Spille, talks about the audio part (7/8 94, p. 364 ff).
And for further informations, please feel free to contact layer3@iis.fhg.de.
3. Applications
Q: O.K., let us concentrate on one or two audio channels. Which Layer shall I
use for my application?
A: Good Question. Of course, it depends on all your requirements. But as a
first approach, you should consider the available bitrate of your
application as the Layers have been designed to support certain areas of
bitrates most effectively. Roughly, today you can achieve a data reduction
of around
1:4 with Layer-1 (or 192 kbps per audio channel),
1:6..8 with Layer-2 (or 128..96 kbps per audio channel), and
1:10..12 with Layer-3, (or 64..56 kbps per audio channel),
and still the reconstructed audio signal will maintain a "CD-like" sound
quality. This may be used as a first "thumb rule" - let's talk about details
later on.
Q: Why does the performance increase with the number of the Layer? Why
does the standard define a family of audio codecs instead of one single
powerful algorithm?
A: Well, the MPEG standard has forged together two main coding schemes
that offered advantages either in complexity (MUSICAM) or in
performance (ASPEC).
Layer-2 is identical with the MUSICAM format. It has been designed as a
trade-off between sound quality per bitrate and encoder complexity. So it is
most useful for the "medium" range of bitrates (96..128 kbps per channel).
For higher bitrates, even a simplified version, the Layer-1, performs well
enough. Layer-1 has originally been developed for a target bitrate of 192
kbps per channel. It is used as "PASC" within the DCC recorder.
For lower bitrates (64 kbps per channel or even less), the Layer-2 format
suffers from its build-in limitations, and with decreasing bitrate, artefacts
become audible more and more. Here is the strong domain of the most
powerful MPEG audio format, Layer-3. It specifies a set of unique features
that all address one goal: to preserve as much sound quality as possible
even at very low bitrates.
Q: Wait a second! I understand that Layer-3 has been an important asset to
the MPEG-1 standard, to address the high-quality low bitrate
applications. With the advent of the "low sample rate extension (LSF)" in
MPEG-2, is it still necessary to rely on Layer-3 to achieve a high-quality
sound at low bitrates?
A: Yes, for sure! Please, don't mix up MPEG-1 and MPEG-2 LSF. MPEG-2
LSF is useful only for applications with limited bandwidth (11.25 kHz, at
best). For applications with full bandwidth, MPEG-1 Layer-3 at 64 or 56
kbps per channel achieves the best sound quality of all ISO codecs.
For applications with limited bandwidth, MPEG-2 LSF Layer-3 provides
an excellent sound quality at 56 kbps for monophonic speech signals and
still a good sound quality at only 64 kbps total bitrate for stereo music
signals (with around 10 kHz bandwidth). The latest MPEG ISO listening
test (in September 94 at NTT Japan, doc. MPEG 94/437) proved the
superior performance of Layer-3 in MPEG-1 and MPEG-2 LSF.
Q: 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: Is there really no alternative to listening tests?
A: No, there is not. With perceptual codecs, all traditional "quality"
parameters (like SNR, THD+N, bandwidth) are rather useless, as any
codec may introduce noise and distortions as long as it does not affect the
perceived sound quality. So, listening tests are necessary, and, if carefully
prepared and performed, lead to rather reliable results.
Nevertheless, Fraunhofer-IIS works on objective sound quality assessment
tools, too. There is already a first product available, the NMR meter, a
real-time DSP-based measurement tool that nicely supports the analysis of
perceptual audio codecs. If you need more informations about the Noise-to-
Mask-Ratio (NMR) technology, feel free to contact nmr@iis.fhg.de.
Q: O.K., back to these listening tests. Come on, tell me some results.
A: Well, for details you should study one of those AES papers or MPEG
documents listed above. The main result is that for low bitrates (64 kbps
per channel or below), Layer-3 always scored significantly better than
Layer-2. Another important conclusion is the draft recommendation of the
task group TG 10/2 within the ITU-R. It recommends the use of low bit-
rate audio coding schemes for digital sound-broadcasting applications
(doc. BS.1115).
Q: Very interesting! Tell me more about this recommendation!
A: The task group TG 10/2 concluded its work in October 93. The draft
recommendation defines three fields of broadcast applications:
- distribution and contribution links (20 kHz bandwidth, no audible
impairments with up to 5 cascaded codecs)
Recommendation: Layer-2 with 180 kbps per channel
- emission (20 kHz bandwidth)
Recommendation: Layer-2 with 128 kbps per channel
- commentary links (15 kHz bandwidth)
Recommendation: Layer-3 with 60 kbps for monophonic and 120 kbps
for stereophonic signals
Q: I see. Medium bitrates - Layer-2, low bitrates - Layer-3. What's about a
bitrate of 96 kbps per channel that seems to be "somewhere in between"
Layer-2 and Layer-3 domains?
A: Interesting question. In fact, a total bitrate of 192 kbps for stereo music is
useful for real applications, e.g. emission via satellite channels. The ITU-R
required that emission codecs should score at least 4.0 on the CCIR
impairment scale, even for the most critical material. At 128 kbps per
channel, Dolby's AC-2, Layer-2 and Layer-3 fulfilled this requirement.
Finally, Layer-2 got the recommendation mainly because of its
"commonality with the distribution and contribution application".
Further tests for emission were performed at 192 kbps joint-stereo coding.
Layer-3 clearly met the requirements, Layer-2 fulfilled them only
marginally, with doubts remaining during further tests with cascaded
codecs in 1993. In the end, the task group decided to pronounce no
recommendation for emission at 192 kbps.
Q: Someone told me that in the ITU-R tests, there was some trouble with
Layer-3, specifically on male voice in the German language. Still, Layer-3
got the recommendation for "commentary links". Can you explain that?
A: Yes. For commentary links, the quality requirements for speech were to be
equivalent to 14-bit linear PCM, and for music, some perceptible
impairments were to be tolerated. In the test in 1992, Layer-3 was by far
the only codec that fulfilled these requirements (e.g. overall monophonic,
Layer-3 scored 3.6 in contrast to Layer-2 at 2.05 - and for male German
speech, Layer-3 scored 4.4 in contrast to Layer-2 at 2.4).
Further tests were performed in 1993 using headphones. They showed that
MPEG-1 Layer-3 with monophonic speech (the test item is German male
voice) at 60 kbps did not fully meet the quality requirements. The ITU
decided to recommend Layer-3 and to include a temporary footnote that
will be removed as soon as an improved Layer-3 codec fulfills the
requirements completely, i.e. even with that well-known critical male
German speech item (for many other speech items, Layer-3 has no trouble
at all).
Q: O.K., a Layer-2 codec at low bitrates may sound poor today, but couldn't
that be improved in the future? I guess you just told me before that the
encoder is not fixed in the standard.
A: Good thinking! As the sound quality mainly depends on the encoder
implementation, it is true that there is no such thing as a "Layer-N"-
quality. So we definitely only know the performance of the reference
codecs used during the international tests. Who knows what will happen in
the future? What we do know now, is:
Today, in MPEG-1 and MPEG-2, Layer-3 provides the best sound quality
at low bitrates, by far better than Layer-2.
Tomorrow, both Layers may improve. Layer-2 has been designed as a
trade-off between quality and complexity, so the bitstream format allows
only limited innovations. In contrast, even the current reference Layer-3-
codec does not exploit all of the powerful mechanisms inside the Layer-3
bitstream format.
Q: What other topics do I have to keep in mind? Tell me about the complexity
of Layer-3.
A: O.K. First, we have to separate between decoder and encoder, as the
workload is distributed asymmetrically between them, i.e. the encoder
needs much more computation power than the decoder.
For a stereo Layer-3-decoder, you may either use a DSP (e.g. one
DSP56002 from Motorola) or an "ASIC", like the masc-programmed DSP
chip MAS 3503 C from Intermetall, ITT. Some rough requirements are:
computation power around 12 MIPs
Data ROM 2.5 Kwords
Data RAM 4.5 Kwords
Programm ROM 2 to 4 Kwords
word length at least 20 bit
Intermetall (ITT) estimated an overhead of around 30 % chip area for
adding the necessary Layer-3 modules to a Layer-2-decoder. So you need
not worry too much about decoder complexity.
For a stereo Layer-3-encoder achieving reference quality, our current real-
time implementations use two DSP32C (AT&T) and one DSP56002. With
the advent of the 21060 (Analog Devices), even a single-chip stereo
encoder comes into view.
Q: Quality, complexity - what about the codec delay?
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)
The practical values are significantly above that. As they depend on the
implementation, exact figures are hard to give. So the figures in brackets
are just rough thumb values - real codecs may show significant higher
values.
Q: For some applications, a very short delay is of critical importance: e.g. in a
feedback link, a reporter can only talk intelligibly if the overall delay is
below around 10 ms. Here, do I have to forget about MPEG audio at all?
A: Not necessarily. In this application, broadcasters may use "N-1" switches
in the studio to overcome this problem - or they may use equipment with
appropriate echo-cancellers.
But with many applications, these delay figures are small enough to
present no extra problem. At least, if one can accept a Layer-2 delay, one
can most likely also accept the higher Layer-3 delay.
Q: Someone told me that, with Layer-3, the codec delay would depend on the
actual audio signal, varying over the time. Is this really true?
A: No. The codec delay does not depend on the audio signal.With all Layers,
the delay depends on the actual implementation used in a specific codec, so
different codecs may have different delays. Furthermore, the delay depends
on the actual sample rate and bitrate of your codec.
Q: All in all, you sound as if anybody should use Layer-3 for low bitrates.
Why on earth do some vendors still offer only Layer-2 equipment for these
applications?
A: Well, maybe because they started to design and develop their systems
rather early, e.g. in 1990. As Layer-2 is identical with MUSICAM, it has
been available since summer of 1990, at latest. In that year, Layer-3
development started and could be successfully finished at the end of 1991.
So, for a certain time, vendors could only exploit the already existing part
of the new MPEG standard.
Now the situation has changed. All Layers are available, the standard is
completed, and new systems may capitalize on the full features of MPEG
audio.
4. Products
Q: What are the main fields of application for Layer-3?
A: Simply put: all applications that need high-quality sound at very low
bitrates to store or transmit music signals. Some examples are:
- high-quality music links via ISDN phone lines (basic rate)
- sound broadcasting via low bitrate satellite channels
- music distribution in computer networks with low demands for channel
bandwidth and memory capacity
- music memories for solid state recorders based on ROM chips
Q: What kind of Layer-3 products are already available?
A: An increasing number of applications benefit from the advanced features
of MPEG audio Layer-3. Here is a list of companies that currently sell
Layer-3 products. For further informations, please contact these companies
directly.
Layer-3 Codecs for Telecommunication:
- AETA, 361 Avenue du Gal de Gaulle (*)
F-92140 Clamart, France
Fax: +33-1-4136-1213 (Mr. Fric)
(*) products announced for 1995
- Dialog 4 System Engineering GmbH, Monreposstr. 57
D-71634 Ludwigsburg, Germany
Fax: +49-7141-22667 (Mr. Burkhardtsmaier)
- PKI Philips Kommunikations Industrie, Thurn-und-Taxis-Str. 14
D-90411 Nuernberg, Germany
Fax: +49-911-526-3795 (Mr. Konrad)
- Telos Systems, 2101 Superior Avenue
Cleveland, OH 44114, USA
Fax: +1-216-241-4103 (Mr. Church)
Speech Announcement Systems:
- Meister Electronic GmbH, Koelner Str. 37
D-51149 Koeln, Germany
Fax: +49-2203-1701-30 (Mr. Seifert)
PC Cards (Hardware and/or Software):
- Dialog 4 System Engineering GmbH, Monreposstr. 57
D-71634 Ludwigsburg, Germany
Fax: +49-7141-22667 (Mr. Burkhardtsmaier)
- Proton Data, Marrensdamm 12 b
D-24944 Flensburg, Germany
Fax: +49-461-38169 (Mr. Nissen)
Layer-3-Decoder-Chips:
- ITT Intermetall GmbH, Hans-Bunte-Str. 19
D-79108 Freiburg, Germany
Fax: +49-761-517-2395 (Mrs. Mayer)
Layer-3 Shareware Encoder/Decoder:
- Mailbox System Nuernberg (MSN), Innerer Kleinreuther Weg 21
D-90408 Nuernberg, Germany
Fax: +49-911-9933661 (Mr. Hanft)
Shareware (version 1.50) is available for:
- IBM-PCs or Compatibles with MS-DOS:
L3ENC.EXE and L3DEC.EXE should work on practically
any PC with 386 type CPU or better. For the encoder, a
486DX33 or better is recommended.
On a 486DX2/66 the current shareware decoder performs in
1:3 real-time, and the shareware encoder in 1:14 real-time
(with stereo signals sampled with 44.1 kHz).
- Sun workstations:
On a SPARC station 10, the decoder works in real time, the
encoder performs in 1:5 real-time.
For more information, refer to chapter 6.
5. Support by Fraunhofer-IIS
Q: I understand that Fraunhofer-IIS has been the main developer of MPEG
audio Layer-3. What can they do for me?
A: The Fraunhofer-IIS focusses on applied research. Its engineers have
profound expertise in real-time implementations of signal-processing
algorithms, especially of Layer-3. The IIS may support a specific Layer-3
application in various ways:
- detailed informations
- technical consulting
- advanced C sources for encoder and decoder
- training-on-the-job
- research and development projects on contract basis.
For more informations, feel free to contact:
- Fraunhofer-IIS, Weichselgarten 3
D-91058 Erlangen, Germany
Fax: +49-9131-776-399 (Mr. Popp)
Q: What are the latest audio demonstrations disclosed by Fraunhofer-IIS?
A: At the Tonmeistertagung 11.94 in Karlsruhe, Germany, the IIS
demonstrated:
- real-time Layer-3 decoder software (mono, 32 kHz fs) including sound
output on ProAudioSpectrum running on a 486DX2/66
- playback of Layer-3 stereo files from a CD-ROM that has been produced
by Intermetall and contains Layer-3 data of up to 15 h of stereo music
(among others, all Beethoven symphonies); the decoder is a small board
that is connected to the parallel printer port. It mainly carries 3 chips: a
PLD as data interface, the MAS 3503 C stereo decoder chip, and the
ASCO Digital-Analog-Converter. The board has two cinch adapters that
allow a very simple connection to the usual stereo amplifier.
- music-from-silicon demonstration by using the standard 1 Mbyte
EPROMs to store 1.5 minutes of CD-like quality stereo music
- music link (with around 6 kHz bandwidth) via V.34 modem at 28.8 kbps
and one analog phone line
6. Shareware Information
The Layer 3 Shareware is copyright Fraunhofer - IIS 1994,1995.
The shareware packages are available:
- via anonymous ftp from fhginfo.fhg.de (153.96.1.4)
You may download our Layer-3 audio software package from the directory
/pub/layer3. You will find the following files:
For IBM PCs:
l3v150d1.txt a short description of the files found in l3v150.zip
l3v150d1.zip encoder, decoder and documentation
l3v150d2.txt a short description of the files found in l3v150n.zip
l3v150d2.zip sample bitstreams
For SUN workstations:
l3v150.sun.txt short description of the files found in
l3v100.sun.tar.gz
l3v150.sun.tar.gz encoder, decoder and documentation
l3v150bit.sun.txt short description of the files found in
l3v150bit.sun.tar.gz
l3v150bit.sun.tar.gz sample bitstreams
- via direct modem download (up to 14.400 bps)
Modem telephone number : +49 911 9933662 Name: FHG
Packet switching network: (0) 262 45 9110 10290 Name: FHG
(For the telephone number, replace "+" with your appropriate international
dial prefix, e.g. "011" for the USA.)
Follow the menus as desired.
- via shipment of diskettes (only including registration)
You may order a diskette directly from:
Mailbox System Nuernberg (MSN)
Hanft & Hartmann
Innerer Kleinreuther Weg 21
D-90408 Nuernberg, Germany
Please note: MSN will only ship a diskette if they get paid for the
registration fee before. The registration fee is 85 Deutsche Mark (about 50
US$) (plus sales tax, if applicable) for one copy of the package. The
preferred method of payment is via credit card. Currently, MSN accepts
VISA, Master Card / Eurocard / Access credit cards. For details see the file
REGISTER.TXT found in the shareware package.
You may reach MSN also via Internet: msn@iis.fhg.de
or via Fax: +49 911 9933661
or via BBS: +49 911 9933662 Name: FHG
or via X25: 0262 45 9110 10290 Name: FHG
(e.g. in USA, please replace "+" with "011"
- via email
You may get our shareware also by a direct request to msn@iis.fhg.de. In
this case, the shareware is split into about 30 small uuencoded parts...
SOFTWARE: MPEG Audio Layer 3 Shareware Codec and Windows Realtime Player
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MPEG Audio Codec and Windows REALTIME Player from Fraunhofer IIS
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Fraunhofer IIS announces l3enc/l3dec V2.00 and WinPlay3 V1.00.
For high quality audio compression, the shareware l3enc/l3dec V2.00
package is available for Linux, SUN, NeXT and DOS on
<URL:ftp://ftp.fhg.de/pub/layer3>
Versions for SGI and HP will follow soon.
The shareware package for DOS
<URL:ftp://ftp.fhg.de/pub/layer3/l3v200d1.zip>
includes a demo version of WinPlay3, a Windows MPEG Audio Layer 3
realtime-player.
With MPEG Audio Layer 3 you can get a 12:1 compression with a CD like
quality.
Instead of 12 MByte / minute (stereo 44.1 kHz) you only need about
1 Mbyte / minute!
More information can be found on
<URL:ftp://ftp.fhg.de/pub/layer3/MPEG_Audio_L3_FAQ.html>
or contact <URL:layer3@iis.fhg.de">mailto:layer3@iis.fhg.de>
- via direct modem download (up to 14.400 bps)
Modem telephone number : +49 911 9933662 Name: FHG
Packet switching network: (0) 262 45 9110 10290 Name: FHG
(For the telephone number, replace "+" with your appropriate international
dial prefix, e.g. "011" for the USA.)
Follow the menus as desired.
- via shipment of diskettes (only including registration)
You may order a diskette directly from:
Mailbox System Nuernberg (MSN)
Hanft & Hartmann
Innerer Kleinreuther Weg 21
D-90408 Nuernberg, Germany
Please note: MSN will only ship a diskette if they get paid for the
registration fee before. The registration fee is 85 Deutsche Mark (about 50
US$) (plus sales tax, if applicable) for one copy of the package. The
preferred method of payment is via credit card. Currently, MSN accepts
VISA, Master Card / Eurocard / Access credit cards. For details see the file
REGISTER.TXT found in the shareware package.
You may reach MSN also via Internet: msn@iis.fhg.de
or via Fax: +49 911 9933661
or via BBS: +49 911 9933662 Name: FHG
or via X25: 0262 45 9110 10290 Name: FHG
(e.g. in USA, please replace "+" with "011"
- via email
You may get our shareware also by a direct request to msn@iis.fhg.de. In
this case, the shareware is split into about 30 small uuencoded parts...
Harald Popp
Audio & Multimedia ("Music is the *BEST*" - F. Zappa)
Fraunhofer-IIS-A, Weichselgarten 3, D-91058 Erlangen, Germany
Phone: +49-9131-776-340
Fax: +49-9131-776-399
email: popp@iis.fhg.de
P.S.: Look out for planetoid #3834!