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Frequently Asked Questions about MPEG Audio Layer-3
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This page gives a comprehensive text about various topics related to audio compression. Because many foreign pages link to this page we decided to maintain it and recommend it as an comprehensive overview for off-line processing as printout. Please send questions and comments to amm_info@iis.fhg.de
This text will be continously upgraded: step by step, more answers and more information will be included. Yes, we definitely know that there are a lot more questions to answer! But we cannot do that all at once. So, some parts may remain "under construction" for a while, and other parts may be modified due to new results of our research work or new applications. You find the latest release at
http://www.iis.fhg.de/departs/amm/layer3/sw/
or
ftp://ftp.fhg.de/pub/iis/layer3/l3faq.html
Introduction - or: What is "MPEG Audio Layer-3"?
Today, efficient coding techniques are a must for cost-effective processing of digital audio and video data by computers. Data reduction of moving pictures and sound is a key technology for any application with limited transmission or storage capacity. In the recent years, a lot of progress has been achieved. While there (still) exist several proprietary formats for audio and video coding, the ISO/IEC standardisation body has released an international standard ("MPEG") for powerful audio and video coding tools (see: Overview about the ISO-MPEG Standard - or: What is MPEG all about?).
Without data reduction, digital audio signals typically consist of 16 bit samples recorded at a sampling rate more than twice the actual audio bandwidth (e.g. 44.1 kHz for Compact Disks). So you end up with more than 1400 kbit to represent just one second of stereo music in CD quality. By using MPEG audio coding, you may shrink down the original sound data from a CD by a factor of 12, without losing sound quality. Factors of 24 and even more still maintain a sound quality that is significantly better than what you get by just reducing the sampling rate and the resolution of your samples. Basically, this is realized by "perceptual coding" techniques addressing the perception of sound waves by the human ear (see: Basics of Perceptual Audio Coding - or: What is the trick?).
Using MPEG audio, one may achieve a typical data reduction of
|
1:4 |
by Layer 1 (corresponds with 384 kbps for a stereo signal), |
|
1:6...1:8 |
by Layer 2 (corresponds with 256..192 kbps for a stereo signal), |
| 1:10...1:12 | by Layer 3 (corresponds with 128..112 kbps for a stereo signal), |
still maintaining the original CD sound quality.
By exploiting stereo effects and by limiting the audio bandwidth, the coding schemes may achieve an acceptable sound quality at even lower bitrates. Layer-3 is the most powerful member of the MPEG audio coding family. For a given sound quality level, it requires the lowest bitrate - or for a given bitrate, it achieves the highest sound quality (see: Advanced Features of Layer-3 - or: Why does Layer-3 perform so well?).
Some typical performance data of Layer-3 are:
| sound quality | bandwidth | mode | bitrate | reduction ratio |
| "telephone sound" | 2.5 kHz | mono | 8 kbps (*) | 96:1 |
| "better than shortwave" | 4.5 kHz | mono | 16 kbps | 48:1 |
| "better than AM radio" | 7.5 kHz | mono | 32 kbps | 24:1 |
| "similar to FM radio" | 11 kHz | stero | 56...64 kbps | 26...24:1 |
| "near-CD" | 15 kHz | stereo | 96 kbps | 16:1 |
| "CD" | >15 kHz | stereo | 112..128kbps | 14..12:1 |
*: Fraunhofer uses a non-ISO extension of Layer-3 for enhanced performance ("MPEG 2.5")
All in all, Layer-3 is the key for numerous low-bitrate, high-quality sound applications (see: Applications - or: Layer-3, what is it good for?).
Applications - or: Layer-3, what is it good for?
A key technology like Layer-3 is useful for a pretty large spectrum of applications - practically almost any system with a limited channel capacity may benefit from it. The following chapters identify some main areas and list some companies that are actively exploiting the Layer-3 technology. For product-related information, please contact these companies directly.
Digital telephone networks (ISDN = Integrated Services Digital Network) offer reliable dial-up links with two 64 kbps data channels per basic rate adapter; other regional networks (in North-America) use 56 kbps data links. Transmission fees are often rather similar or identical to the traditional analog phone lines - those allow to transmit up to 28.8 kbps (V.34 modem) or even 32 kbps ("V.34+").
Using Layer-3, a low-cost narrowband ISDN connection allows to transmit CD-quality sound. Audio professionals, like broadcasting stations and sound studios, benefit from the "music-by-phone" application in various ways. They save money, as they only pay transmission fees for the actual time of usage (not 24 h a day in case of a leased phone line) and for a rather small data channel (one ISDN phone connector for a stereo music link). Radio stations increase the attractiveness of their programs, as reporters transmit high-quality takes (e.g. an interview) or live news without annoying "telephone sound". And new applications become possible, e.g. a "virtual studio", where remote artists may play along some preproduced material, without actually travelling to the studio.
Examples:
Since 1992, various manufacturers are providing equipment ("codecs") for professional audio applications: AVT, Broadcast Electronics, CCS, Dialog 4, Telos.
Pioneered by WorldSpace, a worldwide satellite digital audio broadcasting system is under construction. Its name is "WorldStar", and it will use three geostationary orbit satellites called "AfriStar 1" (21 East), "CaribStar 1" (95 West), and "AsiaStar 1" (105 East), with AfriStar 1 being launched in mid-1998. The other satellites will follow until mid-1999. Each satellite is equipped with three downlink spot beams that are pointed so as to cover populations that provide the greatest radio listener base. Each downlink uses TDM (time division multiplexing) to carry 96 prime rate channels (16 kbps each). The prime rate channels are combined to carry broadcast channels ranging from 16 kbps to 128 kbps; the broadcast channels are coded using MPEG Layer-3. The prime rate channels may even be dynamically allocated to meet the demands of the broadcast service (e.g. 4 channels combined for 1 hour to allow FM quality stereo (64 kbps) for the transmission of a concert with classic music, followed by 1 hour with 4 separate news channels (16 kbps) in 4 different native tongues).
WorldSpace is offering channels on its three satellites for lease to international and national broadcasters. Channel reservation agreements already have been signed with a number of major broadcasters, including Voice of America, Radio Nederland, the Kenya Broadcasting Corporation, the national broadcasting authority of Ghana, the national broadcasting authority of Zimbabwe, New Sky Media of Korea, and RCN of Columbia. Nearly 1 billion $ in private financing has been raised to cover acquisition of the satellites and for most of the operational costs through full system implementation in 1999. France´s Alcatel Espace is the spacecraft prime contractor and supplies the telecommunications payload.
The radio receivers will be designed for maximum convenience of use at a minimum cost. Low cost receiver will use a small compact patch antenna, will require practically no pointing, and will tune automatically to selected channels. Higher end receivers are also envisioned. In a press release from 5. June 96 (Montreux, Switzerland), WorldSpace declared that it has awarded production contracts for two million receiver chips; the contracts were issued to SGS-Thomson and ITT Intermetall, authorizing each company for an initial production of one million receiver chip-sets.
MICRONAS 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.
November 1997:
There is a new chip - "Intermetall MAS3507D" - available, which decodes
Layer-2 and Layer-3. For Layer-3 all features of MPEG-1 and MPEG-2 as well
as Fraunhofer specific - so called "MPEG-2.5" - features for low bit rates
are included. A DC/DC converter for battery-based operation is included.
Power consumption is 90 mW up to a sampling frequency of 24 kHz.
Two housings are available 44-pin PLCC or QFP.
The Internet is a world-wide packet-switched network of computers linked together by various types of data communications systems. Professional Internet providers usually access the network through rather high bit-rate links (e.g., primary rate ISDN with 2 Mbps or ATM with up to 2 Gbps). However, the average consumer uses low cost, low bit-rate connections (e.g., basic rate ISDN with 64 kbps or phone line modems with 28.8 or 14.4 kbps). The actual transmission rate depends on the current user load and the infrastructure of the part of the Internet in use. From a client´s point of view, it may unpredictably vary between zero and the maximum bit-rate of its network modem, with an average bit-rate somewhere in between.
Without audio coding, downloading uncompressed high-quality audio files from a remote Internet server would result in unfavourably long transmission times. For example, with an average transmission rate of 28.8 kbaud (optimistic guess), a single 3-min stereo track from a CD (31.7 Mbyte) would require a download time of more than 2 hours. Therefore, audio on the Internet calls for an audio coding scheme that maintains sound quality as far as possible and allows real-time decoding on a large number of computer platforms without special add-on hardware. Layer-3 fits very well into this scenario - real-time players (like WinPlay3) are available. Intranets present an interesting special case, as they usually provide sufficient bitrate to allow a number of real-time audio links. Furthermore, our experiments indicate that using the http protocol, a real-time connection with 56 (112) kbps is possible with one (two) ISDN phone line(s).
If content providers are willing to add audio data onto their Internet servers, they have to consider carefully the copyright aspects of the music industry (e.g., artists, producers, record companies). They must not violate these rights by their actions! In the framework of a European project called MODE (for "Music-on-Demand"), we developed a flexible protection scheme called MMP (for "MultiMedia protection protocol") that effectively addresses this issue. Furthermore, MMP allows to distribute real-time players "virtually free".
Audio servers may be used plainly for promotional purposes. E.g., museums may increase the attractiveness of their WWW pages by adding some sound files, or mail-order services may add sound excerpts to their server to increase their CD sales numbers. Opticom, a spin-off from Fraunhofer, offers system solutions for this type of application. In spring 1996 (CeBit Hannover), they successfully demonstrated an "audio-on-demand" application via T-Online together with the Deutsche Telekom and a broadcasting station, the Suedwestfunk Baden-Baden.
Another music sales systems has been developed by Cerberus Sound & Vision. The company uses a personalized real-time Layer-3 player and a proprietary encryption scheme to sell sound files via the Internet on a "per song" base. Music servers and mirror sites are currently located in London, New York, Tokyo and Rio; Melbourne and Berlin will follow soon.
"Audio-on-the-Internet" is currently a very popular topic. It does not only comprise audio file transfers with download times as low as possible, but also streaming audio applications, like "Internet Radio". As Layer-3 offers a sound quality "better than shortwave" at a bitrate of 16 kbps (and, with some modifications, may even be useful at 8 kbps), various companies currently work on this Internet subject - e.g., Opticom or Telos.
In a partnership with Apple, Telos introduced in September 96 the Audioactive technology to support "Internet Radio" applications with a live audio input processed by a Layer-3 NetCoder Hardware.
In December 96, Microsoft announced to support MPEG Layer-3 as part of their NetShow multimedia server technology.As first multimedia authoring tools, "Director Multimedia Studio 2" and "SoundEdit 16" (from Macromedia) use Layer-3 to generate compressed sound files for the "shockwave" format.
Layer-3 encoders and decoders are not only available as studio equipment, but also as ISA-bus PC boards from Dialog 4, along with application software, or as low-cost (decoder only) PC boards from NSM; recording and playback tools are also available from Proton Data, along with a special decoder module (called "CenLay3") that allows to playback Layer-3 files via the parallel printer port. Proton Data has also developed a "cutting tool" that allows to manipulate audio data at Layer-3 level.
In addition, a file-oriented Layer-3 encoder and decoder (called "L3ENC" and "L3DEC") is available as shareware for various platforms. Registration is processed by Opticom. Please note that even for registered users, the use of the shareware is limited to "personal edition" purposes.
"WinPlay3" allows the decoding simply by software on any Pentium PC in real time. A 80486 class CPU with a built-in floating-point-unit will also allow some limited operation. For the availability of supported modes, please refer to the following performance matrix:
| Pentium | 486DX4-133 | 486DX2-66 | 486DX-50 | 486DX-33 | |
| MPEG-1 stereo |
ok |
ok |
- | - | - |
| MPEG-1 downmix* |
ok |
ok | ok | - | - |
| MPEG-1 mono |
ok |
ok | ok | ok | - |
| MPEG-2 stereo |
ok |
ok | ok | ok | - |
| MPEG-2 downmix |
ok |
ok | ok | ok | ok |
| MPEG-2 mono |
ok |
ok | ok | ok | ok |
*downmix: the original stereo signal will be played back as a mono signal "MPEG-1" = "MPEG-1 Layer-3", i.e. sample rates 32, 44.1 or 48 kHz "MPEG-2" = "MPEG-2 Layer-3", i.e. sample rates 16, 22.05 or 24 kHz
On a Pentium-90, WinPlay3 consumes less than 30 % of the CPU power to
decode Layer-3 stereo @ 44.1 kHz, or around 5 % of the CPU power to decode
Layer-3 mono @ 16 kHz.
At least, a 8-bit stereo sound card is required. For full quality audio,
a 16-bit card is recommended. The card´s MCI driver should support
sampling frequencies from 8 kHz to 48 kHz.
A standard VGA graphics card is required.
As WinPlay3 buffers up to 4 seconds of sound data due to the limitations
of the Microsoft Windows multitasking architecture, around 1 MByte free
physical memory must be available.
WinPlay3 runs with the following operating systems: Microsoft Windows 3.1/3.11
(in extended 386 mode), Windows 95 und Windows NT (long file names not
yet supported).
WinPlay3 supports file play back of *.mp3 files and direct play from an
URL via HTTP. WinPlay3 can simply be integrated as an helper application
in common browsers, for example Netscape or Mosaic.
WinPlay3 is available at http://www.iis.fhg.de/departs/amm/layer3/winplay3/.
The unregistered player is limited to a reproduction time of 20 sec
(now unlimited V2.3)
, i.e.
it will playback each plain Layer-3 file only for this time. If you want
to use your player without limitation, you have to register your player
with Opticom.
As many applications require a player that is "free" for the user, the latest versions of WinPlay3 (starting with version 2.0) also support the new "MMP" ("MultiMedia protection protocol") format.
MMP is a very flexible data format that may support the following functions:
More detailed information is available at http://www.iis.fhg.de/departs/amm/layer3/mmp/.
In a typical "audio-on-demand" application, the content provider may "on-the-fly" convert its plain Layer-3 data into MMP data, by using a "MMP tagger" software (available at Opticom). The client may use its unregistered player to playback these files without limitation - the player is "virtually free". The client need not pay fees - this issue now may be covered at the server side.
For Mac OS users, a real-time player called "MPEG Layer 3 Player" with a similar look and feel (and similar features) like "WinPlay3" will be released very soon. This new player will (finally!) replace the much simpler (and somewhat buggy) pre-version 0.99 beta that has been available from http://www.iis.fhg.de/departs/amm/layer3/macplay3/.
CD-ROMs (and hard disks) have become most popular to store "multimedia" data. Even with the advent of the new DVD standard, memory capacity will remain a precious resource for many applications. For uncompressed stereo signals from a CD, more than 10 MByte are necessary to store one minute of music. Using Layer-3, less than 1 MByte is enough for the same playing time. And significantly less memory is necessary, if some limitations in performance are acceptable. As CD-ROM readers (and pretty soon, writers too) have already gained a significant market share, typical applications focus today on storing compressed sound files on CD-ROMs, introducing more or better sound tracks into the product. Real application examples are video games, music catalogues or encyclopedias with sound excerpts (e.g., "MusicFinder" by Sygna), or talking books for blind people.
Since fall 96, Bertelsmann is selling their new CD-ROM encyclopedia "Discovery 97" providing information to around 100.000 key words, with rich multimedia information (e.g. more than 2400 coloured photos and images, 41 interactive maps, more than 30 minutes of movie clips, 27 slide shows) including 150 minutes of sound tracks coded with MPEG Layer-3.Up to now, solid-state memories (RAMs, Flash-ROMs) are only used as audio storage devices in special (niche) applications, as the costs per byte are much higher than with other types of media (magneto-optical disks or magnetic tapes). Speech announcement systems for mass transit vehicles (e.g., busses, subways or trains) are an example for such special applications, as the rough environment requires to use ROM based memories. Since 1993, Meister Electronic manufactures speech announcement systems with Layer-3, significantly reducing the precious memory capacity and, at the same time, significantly improving the sound quality (compared with their older 64 kbps PCM "phone sound").
Today, PC-Cards with Flash-ROMs are available, offering a memory capacity up to 100 MByte and more, but at prohibitive high costs for a consumer application. Here, further advances in memory and card technology may trigger a new interesting market segment of "audio-chip-card"-applications. At a press conference in August 95 in Munich, Siemens Germany announced the advent of a new cost-effective ROM technology called the "ROS chip" (ROS = Record-on-Silicon). The first generation of ROS chips will be in production in 1997, with a storage capacity of 64 Mbit; a next generation with 256 Mbit as well as a one-time user programmable version will follow. The ROS chips will be embedded in the new "MultiMedia-Card" from Siemens, a cost-effective card media that will store data, text, graphics, images and sound. Siemens has already demonstrated a battery-powered audio player using a prototype "Audio-Card" containing sound tracks coded with MPEG-Layer-3.
General Questions and Answers
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?
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:
Please note that these papers are not available electronically. You have to order the preprints ("pp. xxxx") directly from the AES.
Addressess
About us - or: What is going on at our Fraunhofer Institute?
FAQ, November 1997, by Harald Popp
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