• 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.

• 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?

• 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.

• 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"

• 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!

• 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.

• 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

• 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&acute;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.

• A: Well, you may contact ISO, or you order it from your national standards body. E.g., in Germany, please contact DIN.

• A: Well, there is "public C source" available on various sites, e.g. at ftp://ftp.fhg.de/pub/iis/layer3/ or at ftp://ftp.tnt.uni-hannover.de/pub/MPEG/audio/mpeg2/public_software/ . This code has been written mainly for explanation purposes, so do not expect too much performance.

Some Basics about MPEG Audio - or: What about Layer-1, Layer-2, Layer-3?

• 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.)

• 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.

• 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).

• 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).

• 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.

• A: Well, more or less, yes - with the exception of the low sample rate extension. Obviously, a pure MPEG-1 decoder is not able to handle the new "half" sample rates.

• 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.

• 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?

• 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

• 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

• 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.

• 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).

• 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).

References - or: Where to find more information?

For around 10 years, perceptual audio coding is a permanent topic at various scientific conferences; e.g., the AES (Audio Engineering Society) organizes two conventions per year. You may find the following papers helpful:

1. Brandenburg, Stoll, et al.: "The ISO/MPEG-Audio Codec: A Generic Standard for Coding of High Quality Digital Audio", 92nd AES, Vienna Mar. 92, pp. 3336; revised version ("ISO-MPEG-1 Audio: A Generic Standard...") published in the Journal of AES, Vol.42, No. 10, Oct. 94
2. Eberlein, Popp, et al.: "Layer-3, a Flexible Coding Standard", 94th AES, Berlin Mar. 93, pp. 3493 3) Church, Grill, et al.: "ISDN and ISO/MPEG Layer-3 Audio Coding: Powerful New tools for Broadcast and Audio Production", 95th AES, New York Oct. 93, pp. 3743
3. Grill, Herre, et al.: "Improved MPEG-2 Audio Multi-Channel Encoding", 96th AES, Amsterdam Feb. 94, pp. 3865
4. Witte, Dietz, et al.: "Single Chip Implementation of an ISO/MPEG Layer-3 Decoder", 96th AES, Amsterdam Feb. 94, pp. 3805
5. Herre, Brandenburg, et al.: "Second Generation ISO/MPEG Audio Layer-3 Coding", 98th AES, Paris Feb. 95
6. Dietz, Popp, et al.: "Audio Compression for Network Transmission", 99th AES, New York Oct. 95, pp. 4129
7. Brandenburg, Bosi: "Overview of MPEG-Audio: Current and Future Standards for Low Bit-Rate Audio Coding, 99th AES, New York Oct. 95, pp. 4130
8. Buchta, Meltzer, et al.: "The WorldStar Sound Format", 101st AES, Los Angeles Nov. 96, pp. 4385
9. Bosi, Brandenburg, et al: "ISO/IEC MPEG-2 Advanced Audio Coding", 101st AES, Los Angeles Nov. 96, pp. 4382

Please note that these papers are not available electronically. You have to order the preprints ("pp. xxxx") directly from the AES.

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, Rathsbergstraße 17
  D-90411 Nürnberg, Germany
  fax: +49 911 5271 100
  contact: Wolfgang Peters
  email: WPeters@avt-nbg.de
  http://www.avt-nbg.de
• Bertelsmann Publishing, Neumarkter Straße 18
  D-81664 München, 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/
• MICRONAS INTERMETALL GmbH, Hans-Bunte-Str. 19
  D-79108 Freiburg, Germany
  fax: +49 761 517 2395
  email: info@intermetall.de
  http://www.intermetall.de/
• Macromedia Inc., 600 Townsend
  San Francisco, CA 94103, USA
  fax: +1 415 626 0554
  http://www.macromedia.com/
• Meister Electronic GmbH, Kölner Str. 37
  D-51149 Köln, 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

What about Fraunhofer Institute?

• 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 Gerhäuser. 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.

• 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.

• 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.

• 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.

Thank you to:

http://www.iis.fhg.de/departs/amm/layers/sw/index.html

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