|Internet media type|
|Type of format||Audio compression format, audio file format|
|Standard|| ISO/IEC 11172-3, |
MPEG-1 Audio Layer II or MPEG-2 Audio Layer II (MP2, sometimes incorrectly called Musicam or MUSICAM) [ citation needed ]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. 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.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
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 finalisation 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 were 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.
CCETT (France), IRT (Germany) and Philips (The Netherlands) won an Emmy Award in Engineering 2000 for development of a digital audio two-channel compression system known as Musicam or MPEG Audio Layer II.
MPEG-1 Audio Layer II is defined in ISO/IEC 11172-3 (MPEG-1 Part 3)
An extension has been provided in MPEG-2 Audio Layer II and is defined in ISO/IEC 13818-3 (MPEG-2 Part 3)
The format is based on successive digital frames of 1152 sampling intervals with four possible formats:
MPEG audio may have variable bit rate (VBR), but it is not widely supported. Layer II can use a method called bit rate switching. Each frame may be created with a different bit rate.According to ISO/IEC 11172-3:1993, Section 184.108.40.206: To provide the smallest possible delay and complexity, the (MPEG audio) decoder is not required to support a continuously variable bit rate when in layer I or II.
Part of the DAB digital radio and DVB digital television standards.
Layer II is commonly used within the broadcast industry for distributing live audio over satellite, ISDN and IP Network connections as well as for storage of audio in digital playout systems. An example is NPR's PRSS Content Depot programming distribution system. The Content Depot distributes MPEG-1 L2 audio in a Broadcast Wave File wrapper. MPEG2 with RIFF headers (used in
.wav) is specified in the RIFF/WAV standards. As a result, Windows Media Player will directly play Content Depot files, however, less intelligent
.wav players often do not. As the encoding and decoding process would have been a significant drain on CPU resources in the first generations of broadcast playout systems, professional broadcast playout systems typically implement the codec in hardware, such as by delegating the task of encoding and decoding to a compatible soundcard rather than the system CPU.
All DVD-Video players in PAL countries contain stereo MP2 decoders, making MP2 a possible competitor to Dolby Digital in these markets. DVD-Video players in NTSC countries are not required to decode MP2 audio, although most do. While some DVD recorders store audio in MP2 and many consumer-authored DVDs use the format, commercial DVDs with MP2 soundtracks are rare.
MPEG-1 Audio Layer II is the standard audio format used in the Video CD and Super Video CD formats (VCD and SVCD also support variable bit rate and MPEG Multichannel as added by MPEG-2).
MPEG-1 Audio Layer II is the standard audio format used in the MHP standard for set-top boxes.
MPEG-1 Audio Layer II is the audio format used in HDV camcorders.
MP2 files are compatible with some Portable audio players.
The term MP2 and filename extension
.mp2 usually refer MPEG-1 Audio Layer II data, but can also refer to MPEG-2 Audio Layer II, a mostly backward compatible extension which adds support for multichannel audio, variable bit rate encoding, and additional sampling rates, defined in ISO/IEC 13818-3. The abbreviation MP2 is also sometimes erroneously applied to MPEG-2 video or MPEG-2 AAC audio.
Sisvel S.p.A., a Luxembourg-based company, administers licenses for patents applying to MPEG Audio.
A codec is a device or computer program which encodes or decodes a data stream or signal. Codec is a portmanteau of coder/decoder.
MP3 is a coding format for digital audio developed largely by the Fraunhofer Society in Germany, with support from other digital scientists in the US and elsewhere. Originally defined as the third audio format of the MPEG-1 standard, it was retained and further extended — defining additional bit-rates and support for more audio channels — as the third audio format of the subsequent MPEG-2 standard. A third version, known as MPEG 2.5 — extended to better support lower bit rates — is commonly implemented, but is not a recognized standard.
MPEG-1 is a standard for lossy compression of video and audio. It is designed to compress VHS-quality raw digital video and CD audio down to about 1.5 Mbit/s without excessive quality loss, making video CDs, digital cable/satellite TV and digital audio broadcasting (DAB) practical.
MPEG-2 is a standard for "the generic coding of moving pictures and associated audio information". It describes a combination of lossy video compression and lossy audio data compression methods, which permit storage and transmission of movies using currently available storage media and transmission bandwidth. While MPEG-2 is not as efficient as newer standards such as H.264/AVC and H.265/HEVC, backwards compatibility with existing hardware and software means it is still widely used, for example in over-the-air digital television broadcasting and in the DVD-Video standard.
Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. Designed to be the successor of the MP3 format, AAC generally achieves higher sound quality than MP3 at the same bit rate.
MPEG-4 Part 3 or MPEG-4 Audio is the third part of the ISO/IEC MPEG-4 international standard developed by Moving Picture Experts Group. It specifies audio coding methods. The first version of ISO/IEC 14496-3 was published in 1999.
Spectral band replication (SBR) is a technology to enhance audio or speech codecs, especially at low bit rates and is based on harmonic redundancy in the frequency domain.
Harmonic Vector Excitation Coding, abbreviated as HVXC is a speech coding algorithm specified in MPEG-4 Part 3 standard for very low bit rate speech coding. HVXC supports bit rates of 2 and 4 kbit/s in the fixed and variable bit rate mode and sampling frequency 8 kHz. It also operates at lower bitrates, such as 1.2 - 1.7 kbit/s, using a variable bit rate technique. The total algorithmic delay for the encoder and decoder is 36 ms.
High-Efficiency Advanced Audio Coding (HE-AAC) is an audio coding format for lossy data compression of digital audio defined as an MPEG-4 Audio profile in ISO/IEC 14496-3. It is an extension of Low Complexity AAC (AAC-LC) optimized for low-bitrate applications such as streaming audio. The usage profile HE-AAC v1 uses spectral band replication (SBR) to enhance the modified discrete cosine transform (MDCT) compression efficiency in the frequency domain. The usage profile HE-AAC v2 couples SBR with Parametric Stereo (PS) to further enhance the compression efficiency of stereo signals.
TwinVQ is an audio compression technique developed by Nippon Telegraph and Telephone Corporation (NTT) Human Interface Laboratories in 1994. The compression technique has been used in both standardized and proprietary designs.
MPEG-4 Part 2, MPEG-4 Visual is a video compression format developed by the Moving Picture Experts Group (MPEG). It belongs to the MPEG-4 ISO/IEC standards. It is a discrete cosine transform (DCT) compression standard, similar to previous standards such as MPEG-1 Part 2 and H.262/MPEG-2 Part 2.
MPEG-4 Audio Lossless Coding, also known as MPEG-4 ALS, is an extension to the MPEG-4 Part 3 audio standard to allow lossless audio compression. The extension was finalized in December 2005 and published as ISO/IEC 14496-3:2005/Amd 2:2006 in 2006. The latest description of MPEG-4 ALS was published as subpart 11 of the MPEG-4 Audio standard in August 2009.
MPEG Multichannel is an extension to the MPEG-1 Layer II audio compression specification, as defined in the MPEG-2 Audio standard which allows it provide up to 5.1-channels of audio. To maintain backwards compatibility with the older 2-channel (stereo) audio specification, it uses a channel matrixing scheme, where the additional channels are mixed into the two backwards compatible channels. Extra information in the data stream contains signals to process extra channels from the matrix.
Packetized Elementary Stream (PES) is a specification in the MPEG-2 Part 1 (Systems) and ITU-T H.222.0 that defines carrying of elementary streams in packets within MPEG program streams and MPEG transport streams. The elementary stream is packetized by encapsulating sequential data bytes from the elementary stream inside PES packet headers.
An elementary stream (ES) as defined by the MPEG communication protocol is usually the output of an audio encoder or video encoder. An ES contains only one kind of data. An elementary stream is often referred to as "elementary", "data", "audio", or "video" bitstreams or streams. The format of the elementary stream depends upon the codec or data carried in the stream, but will often carry a common header when packetized into a packetized elementary stream.
MPEG Surround, also known as Spatial Audio Coding (SAC) is a lossy compression format for surround sound that provides a method for extending mono or stereo audio services to multi-channel audio in a backwards compatible fashion. The total bit rates used for the core and the MPEG Surround data are typically only slightly higher than the bit rates used for coding of the core. MPEG Surround adds a side-information stream to the core bit stream, containing spatial image data. Legacy stereo playback systems will ignore this side-information while players supporting MPEG Surround decoding will output the reconstructed multi-channel audio.
MPEG-1 Audio Layer I, commonly abbreviated to MP1, is one of three audio formats included in the MPEG-1 standard. It is a deliberately simplified version of MPEG-1 Audio Layer II, created for applications where lower compression efficiency could be tolerated in return for a less complex algorithm that could be executed with simpler hardware requirements. While supported by most media players, the codec is considered largely obsolete, and replaced by MP2 or MP3.
Part 3 of the MPEG-2 standard defines audio coding:
Unified Speech and Audio Coding (USAC) is an audio compression format and codec for both music and speech or any mix of speech and audio using very low bit rates between 12 and 64 kbit/s. It was developed by Moving Picture Experts Group (MPEG) and was published as an international standard ISO/IEC 23003-3 and also as an MPEG-4 Audio Object Type in ISO/IEC 14496-3:2009/Amd 3 in 2012.