Informations about MPEG Audio Layer-3

Version 2.00 - 5. 9. 95

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

• layer3@iis.fhg.de
• http://www.iis.fhg.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.

  1. 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
       Monreposstraße 57
       D-71634 Ludwigsburg, Germany
       Fax: +49-7141-22667 (Mr. Burkhardtsmaier)
     • PKI Philips Kommunikations Industrie
       Thurn-und-Taxis-Straße 14
       D-90411 Nürnberg, Germany
       Fax: +49-911-526-3795 (Mr. Konrad)
     • Telos Systems
       2101 Superior Avenue
       Cleveland, OH 44114, USA
       Fax: +1-216-241-4103 (Mr. Church)

  2. Speech Announcement Systems:
     • Meister Electronic GmbH
       Kölner Straße 37
       D-51149 Köln, Germany
       Fax: +49-2203-1701-30 (Mr. Seifert)

  3. PC Cards (Hardware and/or Software):
     • Dialog 4 System Engineering GmbH
       Monreposstraße 57
       D-71634 Ludwigsburg, Germany
       Fax: +49-7141-22667 (Mr. Burkhardtsmaier)
     • Proton Data
       Marrensdamm 12 b
       D-24944 Flensburg, Germany
       Fax: +49-461-3816948 (Mr. Nissen)

  4. Layer-3-Decoder-Chips:
     • ITT Intermetall GmbH
       Hans-Bunte-Straße 19
       D-79108 Freiburg, Germany
       Fax: +49-761-517-2395 (Mrs. Mayer)

  5. Layer-3 Shareware Encoder/Decoder:
     • Opticom, Am Weichselgarten 7
       D-91058 Erlangen, Germany
       Fax: +49-9131-691325 (Mr. Keyhl)
       
       Shareware (current version 2.00) is available for:
       • IBM-PCs or Compatibles with MS-DOS or Linux
       • NeXTSTEP
       • SunOS
       • SGI
       • HP-UX

       The shareware codec works with audio-I/O-files. Non-registered users may only use a limited version. For registration, please contact Opticom.

       Real-time Layer-3 software players (decoder including sound output) are also available; demo-versions are included with the shareware package - they are limited to playback only 20 sec of music.
       For full versions, please contact Opticom.
       The WINDOWS player is called "winplay3". The matrix shows its real-time performance on INTEL machines:

       		    Pentium   486DX2-66  486DX-50   486DX-33		
       MPEG-1 stereo      yes        no         no         no  
       MPEG-1 downmix     yes        yes        no         no
       MPEG-1 mono        yes        yes        yes        no
       MPEG-2 stereo      yes        yes        yes        no
       MPEG-2 downmix     yes        yes        yes        yes
       MPEG-2 mono	   yes        yes        yes        yes 
       
       "downmix" = mono audio output (l+r)/2 from a stereo file input
       

       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)

6. Shareware Information

The Layer 3 Shareware is copyright Fraunhofer - IIS 1994, 1995.

The shareware packages are available via anonymous ftp from ftp.fhg.de

You may download our Layer-3 audio software package from the directory /pub/layer3. You will find the following files:

• For IBM PCs:
  l3v200d1.txt a short description of the files found in l3v200d1.zip
  l3v200d1.zip encoder, decoder and documentation
  l3v200d2.txt a short description of the files found in l3v200d2.zip
  l3v200d2.zip sample bitstreams (same as mp3-bitstreams.tar.gz but as a zip file)
  
• For SUN workstations:
  l3v200.sun.txt short description of the files found in l3v200.sun.tar.gz
  l3v200.sun.tar.gz encoder, decoder and documentation
  mp3-bitstreams.txt short description of the files found in mp3-bitstreams.txt
  mp3-bitstreams.tar.gz sample bitstreams (same as l3v200d2.zip but as a .tar.gz file)
  
• For HP workstations:
  l3v200.hp.txt short description of the files found in l3v200.hp.tar.gz
  l3v200.hp.tar.gz encoder, decoder and documentation
  mp3-bitstreams.txt short description of the files found in mp3-bitstreams.txt
  mp3-bitstreams.tar.gz sample bitstreams (same as l3v200d2.zip but as a .tar.gz file)
  
• For SGI workstations:
  l3v200.sgi.txt short description of the files found in l3v200.sgi.tar.gz
  l3v200.sgi.tar.gz encoder, decoder and documentation
  mp3-bitstreams.txt short description of the files found in mp3-bitstreams.txt
  mp3-bitstreams.tar.gz sample bitstreams (same as l3v200d2.zip but as a .tar.gz file)
  
• For PCs running Linux:
  l3v200.linux.txt short description of the files found in l3v200.linux.tar.gz
  l3v200.linux.tar.gz encoder, decoder and documentation
  mp3-bitstreams.txt short description of the files found in mp3-bitstreams.txt
  mp3-bitstreams.tar.gz sample bitstreams (same as l3v200d2.zip but as a .tar.gz file)
  
• For PCs, Suns, HPs or NeXT-Computer running NeXTSTEP 3.3:
  l3v200.next.txt short description of the files found in l3v200.next.tar.gz
  l3v200.next.tar.gz encoder, decoder and documentation
  mp3-bitstreams.txt short description of the files found in mp3-bitstreams.txt
  mp3-bitstreams.tar.gz sample bitstreams (same as l3v200d2.zip but as a .tar.gz file)
  

* NOTE: You can find lot of them in /OTHER/layer3 folder (OTHER:layer3 for Mac users)

For more information contact layer3@iis.fhg.de, fax to +49-9131-776-399 or send a mail to

FhG-IIS INEL

Weichselgarten 3

D-91058 Erlangen

Germany

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