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Explore the people, places, events, achievements, struggles and stories that shaped our journey.
Movements, leaders, victories and the continuing fight for equality.
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Pioneers, champions, Negro Leagues, records, activism and excellence.
Meet the people whose lives, choices and achievements shaped the journey.
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Wilmington, North Carolina once had a thriving Black middle class and an elected interracial government. In 1898 white supremacists used violence to overthrow that government, kill Black residents and drive many others from the city.
MORE →Reflects the personal views, recollections, and perspective of the author, Mike Davis.
This is a personal recollection on the Move fire on May 13, 1985
An audio coding format,[1] or audio compression format, is an 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.
Audio content encoded in a particular audio coding format is normally encapsulated within a container format. As such, the user normally doesn't have a raw AAC file, but instead has a .m4a audio file, which is a MPEG-4 Part 14 container containing AAC-encoded audio. The container also contains metadata such as title and other tags, and perhaps an index for fast seeking.[2] A notable exception is MP3 files, which are raw audio coding without a container format. De facto standards for adding metadata tags such as title and artist to MP3s, such as ID3, are hacks which work by appending the tags to the MP3, and then relying on the MP3 player to recognize the chunk as malformed audio coding and therefore skip it. In video files with audio, the encoded audio content is bundled with video (in a video coding format) inside a multimedia container format.
An audio coding format does not dictate all algorithms used by a codec implementing the format. An important part of how lossy audio compression works is by removing data in ways humans can't hear, according to a psychoacoustic model; the implementer of an encoder has some freedom of choice in which data to remove (according to their psychoacoustic model).
A lossless audio coding format reduces the total data needed to represent a sound but can be de-coded to its original, uncompressed form. A lossy audio coding format additionally reduces the bit resolution of the sound on top of compression, which results in far less data at the cost of irretrievably lost information.
Transmitted (streamed) audio is most often compressed using lossy audio codecs as the smaller size is far more convenient for distribution. The most widely used audio coding formats are MP3 and Advanced Audio Coding (AAC), both of which are lossy formats based on modified discrete cosine transform (MDCT) and perceptual coding algorithms.
Lossless audio coding formats such as FLAC and Apple Lossless are sometimes available, though at the cost of larger files.
Uncompressed audio formats, such as pulse-code modulation (PCM, or .wav), are also sometimes used. PCM was the standard format for Compact Disc Digital Audio (CDDA).
In 1950, Bell Labs filed the patent on differential pulse-code modulation (DPCM).[3] Adaptive DPCM (ADPCM) was introduced by P. Cummiskey, Nikil S. Jayant and James L. Flanagan at Bell Labs in 1973.[4][5]
Perceptual coding was first used for speech coding compression, with linear predictive coding (LPC).[6] Initial concepts for LPC date back to the work of Fumitada Itakura (Nagoya University) and Shuzo Saito (Nippon Telegraph and Telephone) in 1966.[7] During the 1970s, Bishnu S. Atal and Manfred R. Schroeder at Bell Labs developed a form of LPC called adaptive predictive coding (APC), a perceptual coding algorithm that exploited the masking properties of the human ear, followed in the early 1980s with the code-excited linear prediction (CELP) algorithm which achieved a significant compression ratio for its time.[6] Perceptual coding is used by modern audio compression formats such as MP3[6] and AAC.
Discrete cosine transform (DCT), developed by Nasir Ahmed, T. Natarajan and K. R. Rao in 1974,[8] provided the basis for the modified discrete cosine transform (MDCT) used by modern audio compression formats such as MP3[9] and AAC. MDCT was proposed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987,[10] following earlier work by Princen and Bradley in 1986.[11] The MDCT is used by modern audio compression formats such as Dolby Digital,[12][13] MP3,[9] and Advanced Audio Coding (AAC).[14]
| Basic compression algorithm | Audio coding standard | Abbreviation | Introduction | Market share (2023)[15] | Ref | |
|---|---|---|---|---|---|---|
| Production | Streaming | |||||
| Modified discrete cosine transform (MDCT) | Dolby Digital (AC-3) | AC3 | 1991 | 36–54%[n 1] | 37–61%[n 1] | [12][18] |
| Dolby Digital Plus (E-AC-3) | EAC3 | 2004 | [19][20] | |||
| Adaptive Transform Acoustic Coding | ATRAC | 1992 | Unknown | Unknown | [12] | |
| MPEG Layer III | MP3 | 1993 | 15% | 19% | [9][21] | |
| Advanced Audio Coding (MPEG-2 / MPEG-4) | AAC | 1997 | 83% | 87% | [14][12] | |
| Windows Media Audio | WMA | 1999 | Unknown | Unknown | [12] | |
| Ogg Vorbis | Ogg | 2000 | 6% | 4% | [22][12] | |
| Constrained Energy Lapped Transform | CELT | 2011 | N/a | N/a | [23] | |
| Opus | Opus | 2012 | 12% | 9% | [24] | |
| Dolby AC-4 | AC4 | 2014 | Unknown | Unknown | [25] | |
| LDAC | LDAC | 2015 | Unknown | Unknown | [26][27] | |
| Adaptive differential pulse-code modulation (ADPCM) | aptX / aptX-HD | aptX | 1989 | Unknown | Unknown | [28] |
| Digital Theater Systems | DTS | 1990 | 8% | 6% | [29][30] | |
| Master Quality Authenticated | MQA | 2014 | Unknown | Unknown | ||
| Sub-band coding (SBC) | MPEG-1 Audio Layer II | MP2 | 1993 | Unknown | Unknown | [31] |
| Musepack | MPC | 1997 | ||||
| SBC | SBC | 2003 | Unknown | Unknown | [32] | |
Source: Wikipedia. Article content is retrieved live through the MediaWiki API.
An audio coding format, or audio compression format, is an 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. Audio content encoded in a particular audio coding format is normally encapsulated within a container format. As such, the user normally doesn't have a raw AAC file, but instead has a .m4a audio file, which is a MPEG-4 Part 14 container containing AAC-encoded audio. The container also contains metadata such as title and other tags, and perhaps an index for fast seeking. A notable exception is MP3 files, which are raw audio coding without a container format. De facto standards for adding metadata tags such as title and artist to MP3s, such as ID3, are hacks which work by appending the tags to the MP3, and then relying on the MP3 player to recognize the chunk as malformed audio coding and therefore skip it. In video files with audio, the encoded audio content is bundled with video (in a video coding format) inside a multimedia container format. An audio coding format does not dictate all algorithms used by a codec implementing the format. An important part of how lossy audio compression works is by removing data in ways humans can't hear, according to a psychoacoustic model; the implementer of an encoder has some freedom of choice in which data to remove (according to their psychoacoustic model).
The Internet originated in the efforts of scientists and engineers to build and interconnect computer networks. The Internet Protocol Suite, the set of rules used to communicate between networks and devices on the Internet, arose from research and development in the United States and involved international collaboration, particularly with researchers in the United Kingdom and France. Computer science was an emerging discipline in the late 1950s that began to consider time-sharing between computer users, and later, the possibility of achieving this over wide area networks. J. C. R. Licklider articulated the idea of a universal network at the Information Processing Techniques Office (IPTO) of the United States Department of Defense (DoD) Advanced Research Projects Agency (ARPA). Independently, Paul Baran at the RAND Corporation proposed a distributed network based on data in message blocks in the early 1960s, and Donald Davies conceived of packet switching in 1965 at the National Physical Laboratory (NPL), proposing a national commercial data network in the United Kingdom. ARPA awarded contracts in 1969 for the development of the ARPANET project, directed by Robert Taylor and managed by Lawrence Roberts. ARPANET adopted the packet switching technology proposed by Davies and sought input from Baran. The network of Interface Message Processors (IMPs) was built by a team at Bolt, Beranek, and Newman, with the design and specification led by Bob Kahn. The host-to-host protocol was specified mainly by graduate students, led by Steve Crocker at UCLA, along with Jon Postel and others. The ARPANET expanded rapidly across the United States with connections to the United Kingdom and Norway. Several early packet-switched networks emerged in the 1970s which researched and provided data networking. Louis Pouzin and Hubert Zimmermann pioneered a simplified end-to-end approach to internetworking at the IRIA. Peter Kirstein put internetworking into practice at University College London in 1973. Bob Metcalfe developed the theory and practice behind Ethernet and the PARC Universal Packet. ARPA initiatives and the International Network Working Group developed and refined ideas for internetworking, in which multiple separate networks could be joined into a network of networks. Vint Cerf, now at Stanford University, and Bob Kahn, now at DARPA, published their research on internetworking in 1974. Through the Internet Experiment Note series and later RFCs this evolved into the Transmission Control Protocol (TCP) and Internet Protocol (IP), two protocols of the Internet protocol suite. The design reflected concepts pioneered in the French CYCLADES project directed by Louis Pouzin. The development of packet switching networks was complemented by mathematical work in the 1970s by Leonard Kleinrock at UCLA. In the late 1970s, national and international public data networks emerged based on the X.25 protocol, designed by Rémi Després and others. In the United States, the National Science Foundation (NSF) funded national supercomputing centers at several universities in the United States, and provided interconnectivity in 1986 with the NSFNET project, thus creating network access to these supercomputer sites for research and academic organizations in the United States. International connections to NSFNET, the emergence of architecture such as the Domain Name System, and the adoption of TCP/IP on existing networks in the United States and around the world marked the beginnings of the Internet. Commercial Internet service providers (ISPs) emerged in 1989 in the United States and Australia. Limited private connections to parts of the Internet by officially commercial entities emerged in several American cities by late 1989 and 1990. The optical backbone of the NSFNET was decommissioned in 1995, removing the last restrictions on the use of the Internet to carry commercial traffic, as traffic transitioned to optical networks managed by Sprint, MCI and AT&T in the United States. Research at CERN in Switzerland by the British computer scientist Tim Berners-Lee in 1989–90 resulted in the World Wide Web, linking hypertext documents into an information system, accessible from any node on the network. The dramatic expansion of the capacity of the Internet, enabled by the advent of wave division multiplexing (WDM) and the rollout of fiber optic cables in the mid-1990s, had a revolutionary impact on culture, commerce, and technology. This made possible the rise of near-instant communication by electronic mail, instant messaging, voice over Internet Protocol (VoIP) telephone calls, video chat, and the World Wide Web with its discussion forums, blogs, social networking services, and online shopping sites. Increasing amounts of data are transmitted at higher and higher speeds over fiber-optic networks operating at 1 Gbit/s, 10 Gbit/s, and 800 Gbit/s by 2019. The Internet's takeover of the global communication landscape was rapid in historical terms: it only communicated 1% of the information flowing through two-way telecommunications networks in the year 1993, 51% by 2000, and more than 97% of the telecommunicated information by 2007. The Internet continues to grow, driven by ever greater amounts of online information, commerce, entertainment, and social networking services. However, the future of the global network may be shaped by regional differences.
Internet radio, also known as online radio, web radio, net radio, streaming radio, e-radio and IP radio, is a digital audio service transmitted via the Internet. Broadcasting on the Internet is usually referred to as webcasting since it is not transmitted broadly through wireless means. It can either be used as a stand-alone device running through the Internet, or as a software running through a single computer. Internet radio involves streaming media, presenting listeners with a continuous stream of audio that typically cannot be paused or replayed, much like traditional broadcast media; in this respect, it is distinct from on-demand file serving. Internet radio is also distinct from podcasting, which involves downloading rather than streaming. Internet radio services offer news, sports, talk, and various genres of music—every format that is available on traditional broadcast radio stations. Many Internet radio services are associated with a corresponding traditional (terrestrial) radio station or radio network, although low start-up and ongoing costs have allowed a substantial proliferation of independent Internet-only radio stations. The first Internet radio service was launched in 1993. As of 2017, the most popular Internet radio platforms and applications in the world include TuneIn Radio, iHeartRadio, and SiriusXM. In the U.S., unlike over-the-air broadcast radio, an FCC license is not required to operate an Internet radio service.
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. From the perspective of algebraic signal processing, the relationship between the DCT and the DFT reflects their different underlying algebraic structures: the DFT basis functions are the irreducible representations of the cyclic group, making the DFT the natural spectral transform for signals with periodic (circular) boundary conditions, while the DCT basis functions are the irreducible representations of the dihedral group, making the DCT natural for signals with symmetric (even) boundary conditions. There are eight standard DCT variants, of which four are common. The most common variant of discrete cosine transform is the type-II DCT, which is often called simply the DCT. This was the original DCT as first proposed by Ahmed. Its inverse, the type-III DCT, is correspondingly often called simply the inverse DCT or the IDCT. Two related transforms are the discrete sine transform (DST), which is equivalent to a DFT of real and odd functions, and the modified discrete cosine transform (MDCT), which is based on a DCT of overlapping data. Multidimensional DCTs (MD DCTs) are developed to extend the concept of DCT to multidimensional signals. A variety of fast algorithms have been developed to reduce the computational complexity of implementing DCT. One of these is the integer DCT (IntDCT), an integer approximation of the standard DCT, used in several ISO/IEC and ITU-T international standards. DCT compression, also known as block compression, compresses data in sets of discrete DCT blocks. DCT block sizes include 8 × 8 pixels for the standard DCT, and varied integer DCT sizes between 4 × 4 and 32 × 32 pixels. The DCT has a strong energy compaction property, capable of achieving high quality at high data compression ratios. However, blocky compression artifacts can appear when heavy DCT compression is applied.
Before the 1921 destruction of Tulsa’s Greenwood District, Black residents had created a remarkable center of business and community life. The district included stores, professional offices, entertainment venues and homes owned by Black citizens. Understanding Greenwood means learning what was built—not only what was burned.
MORE →Joe Louis.
