Advanced Audio Coding in the context of Filename extension

Digital television (DTV) is the transmission of television signals using digital encoding, in contrast to the earlier analog television technology which used analog signals. In the 2000s it was represented as the first significant evolution in television technology since color television in the 1950s. Modern digital television is transmitted in high-definition television (HDTV) with greater resolution than analog TV. It typically uses a widescreen aspect ratio (commonly 16:9) in contrast to the narrower format (4:3) of analog TV. It makes more economical use of scarce radio spectrum space; it can transmit up to seven channels in the same bandwidth as a single analog channel, and provides many new features that analog television cannot. A transition from analog to digital broadcasting began around 2000. Different digital television broadcasting standards have been adopted in different parts of the world; below are the more widely used standards:

• Digital Video Broadcasting (DVB) uses coded orthogonal frequency-division multiplexing (OFDM) modulation and supports hierarchical transmission. This standard has been adopted in Europe, Africa, Asia and Australia, for a total of approximately 60 countries.
• Advanced Television System Committee (ATSC) standard uses eight-level vestigial sideband (8VSB) for terrestrial broadcasting. This standard has been adopted by 9 countries: the United States, Canada, Mexico, South Korea, Bahamas, Jamaica, the Dominican Republic, Haiti and Suriname.
• Integrated Services Digital Broadcasting (ISDB) is a system designed to provide good reception to fixed receivers and also portable or mobile receivers utilizing OFDM and two-dimensional interleaving. It supports hierarchical transmission of up to three layers and uses MPEG-2 video and Advanced Audio Coding. This standard has been adopted in Japan and the Philippines. ISDB-T International is an adaptation of this standard using H.264/MPEG-4 AVC, which has been adopted in most of South America as well as Botswana and Angola. 1seg (1-segment) is a special form of ISDB. Each channel is further divided into 13 segments. Twelve are allocated for HDTV and the other for narrow-band receivers such as mobile televisions and cell phones.
• Digital Terrestrial Multimedia Broadcast (DTMB) adopts time-domain synchronous (TDS) OFDM technology with a pseudo-random signal frame to serve as the guard interval (GI) of the OFDM block and the training symbol. The DTMB standard has been adopted in China, including Hong Kong and Macau.
• Digital Multimedia Broadcasting (DMB) is a digital radio transmission technology developed and adopted in South Korea as part of the national information technology project for sending multimedia such as TV, radio and datacasting to mobile devices such as mobile phones, laptops and GPS navigation systems.

A portable media player (PMP) or digital audio player (DAP) is a portable consumer electronics device capable of storing and playing digital media such as audio, images, and video files. Normally, they refer to small, battery-powered devices utilising flash memory or a hard disk for storing various media files. MP3 players has been a popular alternative name used for such devices, even if they also support other file formats and media types other than MP3 (for example AAC, FLAC, WMA).

Generally speaking, PMPs are equipped with a 3.5 mm headphone jack which can be used for headphones or to connect to a boombox, home audio system, or connect to car audio and home stereos wired or via a wireless connection such as Bluetooth, and some may include radio tuners, voice recording and other features. In contrast, analogue portable audio players play music from non-digital media that use analogue media, such as cassette tapes or vinyl records. As devices became more advanced, the PMP term was later introduced to describe players with additional capabilities such as video playback (they used to also be called "MP4 players"). The PMP term has also been used as an umbrella name to describe any portable device for multimedia, including physical formats (such as portable CD players) or handheld game consoles with such capabilities.

An audio coding format (or sometimes audio compression format) is a encoded format of digital audio, such as in digital television, digital radio and in audio and video files. Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.

Some audio coding formats are documented by a detailed technical specification document known as an audio coding specification. Some such specifications are written and approved by standardization organizations as technical standards, and are thus known as an audio coding standard. The term "standard" is also sometimes used for de facto standards as well as formal standards.

A discrete cosine transform (DCT) expresses a finite sequence of data points in terms of a sum of cosine functions oscillating at different frequencies. The DCT, first proposed by Nasir Ahmed in 1972, is a widely used transformation technique in signal processing and data compression. It is used in most digital media, including digital images (such as JPEG and HEIF), digital video (such as MPEG and H.26x), digital audio (such as Dolby Digital, MP3 and AAC), digital television (such as SDTV, HDTV and VOD), digital radio (such as AAC+ and DAB+), and speech coding (such as AAC-LD, Siren and Opus). DCTs are also important to numerous other applications in science and engineering, such as digital signal processing, telecommunication devices, reducing network bandwidth usage, and spectral methods for the numerical solution of partial differential equations.

A DCT is a Fourier-related transform similar to the discrete Fourier transform (DFT), but using only real numbers. The DCTs are generally related to Fourier series coefficients of a periodically and symmetrically extended sequence whereas DFTs are related to Fourier series coefficients of only periodically extended sequences. DCTs are equivalent to DFTs of roughly twice the length, operating on real data with even symmetry (since the Fourier transform of a real and even function is real and even), whereas in some variants the input or output data are shifted by half a sample.

A CD player is an electronic device that plays audio compact discs, which are a digital optical disc data storage format. CD players were first sold to consumers in 1982. CDs typically contain recordings of audio material such as music or audiobooks. CD players may be part of home stereo systems, car audio systems, personal computers, or portable CD players such as CD boomboxes. Most CD players produce an output signal via a headphone jack or RCA jacks. To use a CD player in a home stereo system, the user connects an RCA cable from the RCA jacks to a hi-fi (or other amplifier) and loudspeakers for listening to music. To listen to music using a CD player with a headphone output jack, the user plugs headphones or earphones into the headphone jack.

Modern units can play audio formats other than the original CD PCM audio coding, such as MP3, AAC and WMA. DJs playing dance music at clubs often use specialized players with an adjustable playback speed to alter the pitch and tempo of the music. Audio engineers using CD players to play music for an event through a sound reinforcement system use professional audio-grade CD players. CD playback functionality is also available on CD-ROM/DVD-ROM drive-equipped computers as well as on DVD players and most optical disc-based home video game consoles.

A music download is the digital transfer of music via the Internet into a device capable of decoding and playing it, such as a personal computer, portable media player, MP3 player or smartphone. This term encompasses both legal downloads and downloads of copyrighted material without permission or legal payment. Music downloads are typically encoded with the MP3 audio coding format. or using the modified discrete cosine transform (MDCT) audio data compression, particularly the Advanced Audio Coding (AAC) format used by iTunes.

Since the advent of streaming, downloads as a mode of music distribution has seen a steady decline from its peak in the early 2010s. According to a Nielsen report, downloadable music accounted for 55.9 percent of all music sales in the US in 2012. By the beginning of 2011, Apple's iTunes Store alone made US$1.1 billion of revenue in the first quarter of its fiscal year. According to the RIAA, music downloads peaked at 43% of industry revenue in the US in 2012, and has since fallen to 3% in 2022.

The modified discrete cosine transform (MDCT) is a transform based on the type-IV discrete cosine transform (DCT-IV), with the additional property of being lapped: it is designed to be performed on consecutive blocks of a larger dataset, where subsequent blocks are overlapped so that the last half of one block coincides with the first half of the next block. This overlapping, in addition to the energy-compaction qualities of the DCT, makes the MDCT especially attractive for signal compression applications, since it helps to avoid artifacts stemming from the block boundaries. As a result of these advantages, the MDCT is the most widely used lossy compression technique in audio data compression. It is employed in most modern audio coding standards, including MP3, Dolby Digital (AC-3), Vorbis (Ogg), Windows Media Audio (WMA), ATRAC, Cook, Advanced Audio Coding (AAC), High-Definition Coding (HDC), LDAC, Dolby AC-4, and MPEG-H 3D Audio, as well as speech coding standards such as AAC-LD (LD-MDCT), G.722.1, G.729.1, CELT, and Opus.

The discrete cosine transform (DCT) was first proposed by Nasir Ahmed in 1972, and demonstrated by Ahmed with T. Natarajan and K. R. Rao in 1974. The MDCT was later proposed by John P. Princen, A.W. Johnson and Alan B. Bradley at the University of Surrey in 1987, following earlier work by Princen and Bradley (1986) to develop the MDCT's underlying principle of time-domain aliasing cancellation (TDAC), described below. (There also exists an analogous transform, the MDST, based on the discrete sine transform, as well as other, rarely used, forms of the MDCT based on different types of DCT or DCT/DST combinations.)

High-Efficiency Advanced Audio Coding (HE-AAC) is an audio coding format for lossy data compression of digital audio as part of the MPEG-4 standards. 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.

The MPEG-4 Low Delay Audio Coder (a.k.a. AAC Low Delay, or AAC-LD) is audio compression standard designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It is closely derived from the MPEG-2 Advanced Audio Coding (AAC) standard. It was published in MPEG-4 Audio Version 2 (ISO/IEC 14496-3:1999/Amd 1:2000) and in its later revisions.

AAC-LD uses a version of the modified discrete cosine transform (MDCT) audio coding technique called the LD-MDCT. AAC-LD is widely used by Apple as the voice-over-IP (VoIP) speech codec in FaceTime.

An audio coding format, or audio compression format, is a encoded format of digital audio, such as in digital television, digital radio and in audio and video files. Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.

Some audio coding formats are documented by a detailed technical specification document known as an audio coding specification. Some such specifications are written and approved by standardization organizations as technical standards, and are thus known as an audio coding standard. The term "standard" is also sometimes used for de facto standards as well as formal standards.

Variable bitrate (VBR) is a term used in telecommunications and computing that relates to the bitrate used in sound or video encoding. As opposed to constant bitrate (CBR), VBR files vary the amount of output data per time segment. VBR allows a higher bitrate (and therefore more storage space) to be allocated to the more complex segments of media files while less space is allocated to less complex segments. The average of these rates can be calculated to produce an average bitrate for the file.

MP3, WMA and AAC audio files can optionally be encoded in VBR, while Opus and Vorbis are encoded in VBR by default. Variable bit rate encoding is also commonly used on MPEG-2 video, MPEG-4 Part 2 video (Xvid, DivX, etc.), MPEG-4 Part 10/H.264 video, Theora, Dirac and other video compression formats. Additionally, variable rate encoding is inherent in lossless compression schemes such as FLAC and Apple Lossless.