Iso Iec 11172 3 Pdf Download 2021
MPEG-1 Audio Layer II or MPEG-2 Audio Layer II (MP2, sometimes incorrectly called Musicam or MUSICAM)[7] 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.[8]
Iso Iec 11172 3 Pdf Download
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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.[19]
The MPEG-1 Audio standard was based on the existing MUSICAM and ASPEC audio formats.[21]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[22][23][24] and finalized in 1992[25] 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.[4] Further work on MPEG audio[26] 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[27]), originally published in 1995.[5][28] 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.[26]
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.[32][33]According to ISO/IEC 11172-3:1993, Section 2.4.2.3: 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.[34]
(1) MPEG-1: ISO/IEC 11172-3. Information technology -- Coding of moving pictures and associated audio for digital storage media at up to about 1,5 Mbit/s -- Part 3: Audio. (2) MPEG-2: ISO/IEC 13818-3. Information technology -- Generic coding of moving pictures and associated audio information -- Part 3: Audio.
Specifies the system layer of the coding. Was developed principally to support the combination of the video and audio coding methods defined in ISO/IEC 11172-2 and ISO/IEC 11172-3. The system layer supports the following basic functions: the synchronization of multiple compressed streams on playback, the interleaving of multiple compressed streams into a single stream, the initializiation of buffering for playback start up, continuous buffer management, and time identification.
The MPEG was the first to develop a complete system-level standard solution for digital audio and video. The MPEG-1 standard includes a part for video compression, another for audio compression and a third for the audiovisual stream (Watkinson 2004Watkinson J (2004) The MPEG Handbook. 2nd ed. Waltham: Focal Press.). The MPEG-1 audio standard (ISO 1993aISO (1993a) ISO/IEC 11172-1:1993 Information technology - Coding of moving pictures and associated audio for digital storage media at up to about 1,5 Mbit/s - Part 1: Systems. Geneva: ISO. [accessed Jun 15 2018]. ) describes a perceptual audio coding algorithm designed for general audio signals. This standard had application in areas such as digital compact cassette (DCC), digital audio and video broadcasting (DAB and DVB), and portable devices, among others. As a matter of explanation on the subject, MPEG-1 and MPEG-2 are defined as audio and video coding systems, whereas the layers I (MP1), II (MP2), and III (MP3) of MPEG are associated to audio coding, thus being the body of this study. The MP2, MP3, and advanced audio coding (AAC) audio formats defined by the MPEG group were selected because they are based on perceptual coding format, which means that part of the audio listened by the human ear is discarded following the psychoacoustic model. In practice, the frequency range best perceived by the human ear remains the same up to 4 kHz, and outside this range a quantization factor is applied to reduce the quality of the signal, thus requiring fewer data to be stored.
The MPEG-1 Layer II or MP2 format defined by the ISO/IEC 11172-3:1993 (ISO 1993bISO (1993b) ISO/IEC 11172-3:1993 Information technology - Coding of moving pictures and associated audio for digital storage media at up to about 1,5 Mbit/s - Part 3: Audio. Geneva: ISO. [accessed Jun 15 2018]. ) and ISO/IEC 13818-3:1998 (ISO 1998ISO (1998) ISO/IEC 13818-3:1998 Information technology - Generic coding of moving pictures and associated audio information - Part 3: Audio. Geneva: ISO. [accessed Jun 15 2018]. ) standards as a lossy audio compression format, which had been created according to the MUSICAM audio codec (Dehery et al. 1991Dehery YF, Lever M, Urcun P (1991) A MUSICAM source codec for digital audio broadcasting and storage. Paper presented [Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing. IEEE; Toronto, Canada. ). The MUSICAM audio codec was designed to work in a wide range of applications, from broadcasting and telecommunication, up to digital storage media.
The MPEG-1 Layer III or MP3 format defined by ISO/IEC 11172-3:1993 (ISO 1993bISO (1993b) ISO/IEC 11172-3:1993 Information technology - Coding of moving pictures and associated audio for digital storage media at up to about 1,5 Mbit/s - Part 3: Audio. Geneva: ISO. [accessed Jun 15 2018]. ), is considered to be the most complex layer of this ISO standard. This format is the evolution of the MP2 format with the addition of a hybrid transform, nonuniform quantization with entropy coding, and different window sizes. This has allowed increasing the compression rate from 8:1 to 12:1 (Madisetti 2010Madisetti V (2010) Video, speech, and audio signal processing and associated standards. Boca Raton: CRC Press. ).