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

Version 2.60

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 Franhofer Institute?

Introduction - or: What is "MPEG Audio Layer-3"?

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),

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      stereo    56..64 kbps    26..24:1
"near-CD"                15 kHz      stereo    96 kbps        16:1
"CD"                     > 15 kHz    stereo    112..128 kbps  14..12:1
*: Fraunhofer uses a non-ISO extension of Layer-3 for enhanced performance ("MPEG 2.5")

Applications - or: Layer-3, what is it good for?

Music Links via ISDN

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.

Digital Satellite Broadcasting

WorldSpace is offering channels on its three satellites for lease to international and national broadcasters. Agreements already have been signed with a number of broadcasters, and negotiations are underway with numerous other system users. 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. FranceÂs Alcatel Espace is the spacecraft prime contractor and supplies the telecommunications payload.

The radio receivers (named StarMan”) 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 StarMan 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

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 format") 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 Südwestfunk Baden-Baden.

Audioservers may also be used for music sales systems. Cerberus Sound amp Vision uses a personalized real-time Layer-3 player and a proprietary encryption scheme to sell sound files via the Internet on a " BERLIN WILL

"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. As first multimedia authoring tools, "Director Multimedia Studio 2" and "SoundEdit 16" (from Macromedia) exploit Layer-3 to generate compressed sound files for "Shockwave" movies.

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; 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.

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.

Real-time Layer-3 players

WinPlay3

                Pentium 486DX2-66  486DX-50  486DX-33
MPEG-1 stereo     ok        -          -         -
MPEG-1 downmix*   ok        ok         -         -
MPEG-1 mono       ok        ok         ok        -
MPEG-2 stereo     ok        ok         ok        -
MPEG-2 downmix*   ok        ok         ok        ok
MPEG-2 mono       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

MMP

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

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

Layer-3 Sound on CD-ROMs

Layer-3 Sound on Silicon

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/audio/mpeg2/public_software/ . This code has been written mainly for explanation purposes, so do not expect too much performance.

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

• Q: Talking about MPEG audio, I always hear "Layer 1, 2 and 3". What does it mean?
• A: MPEG describes the compression of audio signals using high performance perceptual coding schemes. It specifies a family of three audio coding schemes, simply called Layer-1, Layer-2, and Layer-3. From Layer-1 to Layer-3, encoder complexity and performance (sound quality per bitrate) are increasing.

The three codecs are compatible in a hierarchical way, i.e. a Layer-N decoder may be able to decode bitstream data encoded in Layer-N and all Layers below N (e.g., a Layer-3 decoder may accept Layer-1,-2,-3, whereas a Layer-2 decoder may accept only Layer-1 and -2.)
• Q: So we have a family of three audio coding schemes. What does the MPEG standard define, exactly?
• A: For each Layer, the standard specifies the bitstream format and the decoder. To allow for future improvements, it does not specify the encoder, but an informative chapter gives an example for an encoder for each Layer.
• Q: What have the three audio Layers in common?
• A: All Layers use the same basic structure. The coding scheme can be described as "perceptual noise shaping" or "perceptual subband / transform coding". The encoder analyzes the spectral components of the audio signal by calculating a filterbank (transform) and applies a psychoacoustic model to estimate the just noticeable noise-level. In its quantization and coding stage, the encoder tries to allocate the available number of data bits in a way to meet both the bitrate and masking requirements.

The decoder is much less complex. Its only task is to synthesize an audio signal out of the coded spectral components.
All Layers use the same analysis filterbank (polyphase with 32 subbands). Layer-3 adds a MDCT transform to increase the frequency resolution.
All Layers use the same "header information" in their bitstream, to support the hierarchical structure of the standard.
All Layers have a similar sensitivity to biterrors. They use a bitstream structure that contains parts that are more sensitive to biterrors ("header", "bit allocation", "scalefactors", "side information") and parts that are less sensitive ("data of spectral components").
All Layers support the insertion of programm-associated information ("ancillary data") into their audio data bitstream.
All Layers may use 32, 44.1 or 48 kHz sampling frequency.
All Layers are allowed to work with similar bitrates:
Layer-1: from 32 kbps to 448 kbps
Layer-2: from 32 kbps to 384 kbps
Layer-3: from 32 kbps to 320 kbps
The last two statements refer to MPEG-1; with MPEG-2, there is an extension for the sampling frequencies and bitrates (see below).
• Q: What are the main differences between the three Layers, from a global view?
• A: From Layer-1 to Layer-3, complexity increases (mainly true for the encoder), overall codec delay increases, and performance increases (sound quality per bitrate).
• Q: What are the main differences between MPEG-1 and MPEG-2 in the audio part?
• A: MPEG-1 and MPEG-2 use the same family of audio codecs, Layer-1, -2 and -3. The new audio features of MPEG-2 are a "low sample rate extension" to address very low bitrate applications with limited bandwidth requirements (the new sampling frequencies are 16, 22.05 or 24 kHz, the bitrates extend down to 8 kbps), and a "multichannel extension" to address surround sound applications with up to 5 main audio channels (left, center, right, left surround, right surround) and optionally 1 extra "low frequency enhancement (LFE)" channel for subwoofer signals; in addition, a "multilingual extension" allows the inclusion of up to 7 more audio channels.
• Q: Is this all compatible to each other?
• A: Well, more or less, yes - with the execption of the low sample rate extension. Obviously, a pure MPEG-1 decoder is not able to handle the new "half" sample rates.
• Q: You mean: compatible!? With all these extra audio channels? Please explain!
• A: Compatibility has been a major topic during the MPEG-2 definition phase. The main idea is to use the same basic bitstream format as defined in MPEG-1, with the main data field carrying two audio signals (called L0 and R0) as before, and the ancillary data field carrying the multichannel extension information. Without going further into details, two terms should be explained here: "forwards compatible": the MPEG-2 decoder has to accept any MPEG-1 audio bitstream (that represents one or two audio channels) "backwards compatible": the MPEG-1 decoder should be able to decode the audio signals in the main data field (L0 and R0) of the MPEG-2 bitstream "Matrixing" may be used to get the surround information into L0 and R0: L0 = left signal + a * center signal + b * left surround signal R0 = right signal + a * center signal + b * right surround signal Therefore, a MPEG-1 decoder can reproduce a comprehensive downmix of the full 5- channel information. A MPEG-2 decoder uses the multichannel extension information (3 more audio signals) to reconstruct the five surround channels.
• Q: In your footnotes, you indicate the use of some "non-ISO" extension inside your Fraunhofer codec, called "MPEG 2.5", to further improve the performance at very low bitrates (e.g. 8 kbps mono). What do you mean by this?
• A: Oh, yes. Well, the MPEG-2 standard allows bitrates as low as 8 kbps, for the low sample rate extension. At such a low bitrate, the useful audio bandwidth has to be limited anyway, e.g. to 3 kHz. Therefore, the actual sample rate could be reduced, e.g. to 8 kHz. The lower the sample rate, the better the frequency resolution, the worse the time resolution, and the better the ratio between control information and audio payload inside the bitstream format. As the MPEG-2 standard defines 16 kHz as lowest sample rate, we introduced a further extension, again dividing the low sample rates of MPEG-2 by 2, i.e. we introduced 8, 11.025, and 12 kHz - and we named this extension to the extension "MPEG 2.5". "Layer-3" performs significantly better with 8 kbps @ 8 kHz or 16 kbps @ 11 kHz than with 8 or 16 kbps @ 16 kHz.

Advanced Features of Layer-3 - or: Why does Layer-3 perform so well?

• Q: Well, I read your statement about "CD-like" performance, achieved at a data reduction of 4:1 (or 384 kbps total bitrate) with Layer-1, 6..8:1 (or 256..192 kbps total bitrate) with Layer-2, and 12..14:1 (or 128..112 kbps total bitrate) with Layer-3. Can you explain a little further?
• A: Well, each audio Layer extends the features of the Layer with the lower number. The simplest form is Layer-1. It has been designed mainly for the DCC (Digital Compact Cassette), where it is used at 384 kbps (called "PASC"). Layer-2 has been designed as a trade-off between complexity and performance. It achieves a good sound quality at bitrates down to 192 kbps. Below, sound quality suffers. Layer-3 has been designed for low bitrates right from the start. It adds a number of "advanced features" to Layer-2: the frequency resolution is 18 times higher, which allows a Layer-3 encoder to adapt the quantisation noise much better to the masking threshold only Layer-3 uses entropy coding (like MPEG video) to further reduce redundancy only Layer-3 uses a bit reservoir (like MPEG video) to suppress artefacts in critical moments and Layer-3 may use more advanced joint-stereo coding methods
• Q: I see. Sounds to me as if Layer-3 is something like a "Layer-2++". Now, tell me more about sound quality. How do you assess that?
• A: Today, there is no alternative to expensive listening tests. During the ISO-MPEG process, a number of international listening tests have been performed, with a lot of trained listeners. All these tests used the "triple stimulus, hidden reference" method and the "CCIR impairment scale" to assess the sound quality. The listening sequence is "ABC", with A = original, BC = pair of original / coded signal with random sequence, and the listener has to evaluate both B and C with a number between 1.0 and 5.0. The meaning of these values is: 5.0 = transparent (this should be the original signal) 4.0 = perceptible, but not annoying (first differences noticable) 3.0 = slightly annoying 2.0 = annoying 1.0 = very annoying
• Q: Listening tests are certainly an expensive task. Is there really no alternative?
• A: Well, at least not today. Tomorrow may be different. To assess sound quality with perceptual codecs, all traditional "quality" parameters (like signal-to-noise ratio, total harmonic distortion, bandwidth) are rather useless, as any codec may introduce noise and distortions as long as these do not affect the perceived sound quality. So, listening tests are necessary, and, if carefully prepared and performed, they lead to rather reliable results.

Nevertheless, Fraunhofer-IIS works on the development and standardisation of objective sound quality assessment tools, too. And there is already a first product available (contact Opticom), a real-time measurement tool that nicely supports the analysis of perceptual audio codecs. If you need more information about the Noise- to-Mask-Ratio (NMR) technology, feel free to contact nmr@iis.fhg.de.
• Q: O.K., back to these listening tests and the performance evaluation. Come on, tell me some results.
• A: Well, for more details you should study one of these AES papers or the MPEG documents. For Layer-3, the main result is that it always performed superior at low bitrates (64 kbps per audio channel or below). Well, this is not completely surprising, as Layer-3 uses the same tool set as Layer-2, but with some additional advanced coding features that all address the demands of very low bitrate coding. One impressive example is the ISO-MPEG listening test carried out in September 94 at NTT Japan (doc. ISO/IEC JTC1/SC29/WG11 N0848, 11.Nov. 94). Another interesting result is the conclusion of the task group TG 10/2 within the ITU- R, which recommends the use of low bit-rate audio coding schemes for digital sound-broadcasting applications (ITU-R doc. BS.1115).
• Q: Very interesting! Tell me more about this recommendation!
• A: The task group TG 10/2 finished its work in 10/93. The recommendation defines three fields of broadcast applications and recommends Layer-2 with 180 kbps per channel for distribution and contribution links (20 kHz bandwidth, no audible impairments with up to 5 cascaded codec), Layer-2 with 128 kbps per channel for emission (20 kHz bandwidth), and Layer-3 with 60 (120) kbps for mono (stereo) signals for commentary links (15 kHz bandwidth).

Basics of Perceptual Audio Coding - or: What is the trick?

References - or: Where to find more information?

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

Addressess

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 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 skilled 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. In any case, 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.