MPEG-1 Audio Layer II or MPEG-2 Audio Layer II (MP2, sometimes incorrectly called Musicam or MUSICAM) is a lossy audio compression format defined by ISO/IEC 11172-3 alongside MPEG-1 Audio Layer I and MPEG-1 Audio Layer III (MP3). While MP3 is much more popular for PC and Internet applications, MP2 remains a dominant standard for audio broadcasting.
MPEG-1 Audio Layer 2 encoding was derived from the MUSICAM (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) audio codec, developed by Centre commun d'études de télévision et télécommunications (CCETT), Philips, and the Institut für Rundfunktechnik (IRT) in 1989 as part of the EUREKA 147 pan-European inter-governmental research and development initiative for the development of a system for the broadcasting of audio and data to fixed, portable or mobile receivers (established in 1987).
It began as the Digital Audio Broadcast (DAB) project managed by Egon Meier-Engelen of the Deutsche Forschungs- und Versuchsanstalt für Luft- und Raumfahrt (later on called Deutsches Zentrum für Luft- und Raumfahrt, German Aerospace Center) in Germany. The European Community financed this project, commonly known as EU-147, from 1987 to 1994 as a part of the EUREKA research program.
The Eureka 147 System comprised three main elements: MUSICAM Audio Coding (Masking pattern Universal Sub-band Integrated Coding And Multiplexing), Transmission Coding & Multiplexing and COFDM Modulation.
MUSICAM was one of the few codecs able to achieve high audio quality at bit rates in the range of 64 to 192 kbit/s per monophonic channel. It has been designed to meet the technical requirements of most applications (in the field of broadcasting, telecommunication, and recording on digital storage media) — low delay, low complexity, error robustness, short access units, etc.
As a predecessor of the MP3 format and technology, the perceptual codec MUSICAM is based on integer arithmetics 32 subbands transform, driven by a psychoacoustic model. It was primarily designed for Digital Audio Broadcasting and digital TV, and disclosed by CCETT(France) and IRT (Germany) in Atlanta during an IEEE-ICASSP conference. This codec incorporated into a broadcasting system using COFDM modulation was demonstrated on air and on the field together with Radio Canada and CRC Canada during the NAB show (Las Vegas) in 1991. The implementation of the audio part of this broadcasting system was based on a two chips encoder (one for the subband transform, one for the psychoacoustic model designed by the team of G. Stoll (IRT Germany), later known as Psychoacoustic model I in the ISO MPEG audio standard) and a real-time decoder using one Motorola 56001 DSP chip running an integer arithmetics software designed by Y.F. Dehery's team (CCETT, France). The simplicity of the corresponding decoder together with the high audio quality of this codec using for the first time a 48 kHz sampling frequency, a 20 bits/sample input format (the highest available sampling standard in 1991, compatible with the AES/EBU professional digital input studio standard) were the main reasons to later adopt the characteristics of MUSICAM as the basic features for an advanced digital music compression codec such as MP3.
The audio coding algorithm used by the Eureka 147 Digital Audio Broadcasting (DAB) system has been subject to the standardization process within the ISO/Moving Pictures Expert Group (MPEG) in 1989–94. MUSICAM audio coding was used as a basis for some coding schemes of MPEG-1 and MPEG-2 Audio. Most key features of MPEG-1 Audio were directly inherited from MUSICAM, including the filter bank, time-domain processing, audio frame sizes, etc. However, improvements were made, and the actual MUSICAM algorithm was not used in the final MPEG-1 Audio Layer II standard.
Since the finalization of MPEG-1 Audio and MPEG-2 Audio (in 1992 and 1994), the original MUSICAM algorithm is not used anymore. The name MUSICAM is often mistakenly used when MPEG-1 Audio Layer II is meant. This can lead to some confusion, because the name MUSICAM is trademarked by different companies in different regions of the world. (Musicam is the name used for MP2 in some specifications for Astra Digital Radio as well as in the BBC's DAB documents.)
The Eureka Project 147 resulted in the publication of European Standard, ETS 300 401 in 1995, for DAB which now has worldwide acceptance. The DAB standard uses the MPEG-1 Audio Layer II (ISO/IEC 11172-3) for 48 kHz sampling frequency and the MPEG-2 Audio Layer II (ISO/IEC 13818-3) for 24 kHz sampling frequency.
In the late 1980s, ISO's Moving Picture Experts Group (MPEG) started an effort to standardize digital audio and video encoding, expected to have a wide range of applications in digital radio and TV broadcasting (later DAB, DMB, DVB), and use on CD-ROM (later Video CD). The MUSICAM audio coding was one of 14 proposals for MPEG-1 Audio standard that was submitted to ISO in 1989.
The MPEG-1 Audio standard was based on the existing MUSICAM and ASPEC audio formats. The MPEG-1 Audio standard included the three audio "layers" (encoding techniques) now known as Layer I (MP1), Layer II (MP2) and Layer III (MP3). All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 as the committee draft of ISO-11172 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993. Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Part 3 or backward-compatible MPEG-2 Audio or MPEG-2 Audio BC), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II, and III. The new sampling rates are exactly half that of those originally defined for MPEG-1 Audio. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.
The Layer III (MP3) component uses a lossy compression algorithm that was designed to greatly reduce the amount of data required to represent an audio recording and sound like a decent reproduction of the original uncompressed audio for most listeners.