MP3

After that, MP3 algorithms become public domain and all of these counter-productive legal hassles go away. See also: Jacob Nielsen on advanced hypertext for the World Wide Web. The Fraunhofer patents expire April 2010. For instance, members of the modern social software landscape such as the wiki web and nukes allow surfers to edit the web pages they visit. With Thomson and Sivel both owning separate patents which they claim are needed by the codec, the legal status of MP3 remains unclear. Sometimes web services or browser manufacturers remedy these shortcomings. Motorola also recently signed with Audio MPEG to license MP3. Even some hypertext features that were in early versions of HTML have been ignored by most popular webbrowsers until now, such as the link element and editable webpages.

[3] previously sued Thomson for patent infringement on MP3 technology[4], but those disputes were resolved in November 2005 with Sisvel granting Thomson an MP3 license. Another feature lacking today are fat links. [2] and its US subsidiary Audio MPEG, Inc. Earlier hypertext systems had features such as typed links, transclusion and source tracking. Sisvel S.p.A. HTML is the basis of a comparatively weak hypertext implementation. Thus while patent fees have been an issue for companies attempting to use MP3, they have not meaningfully impacted users, allowing the format to grow in popularity. For these reasons many mailing lists deliberately block HTML email either stripping out the HTML part to just leave the plain text part or rejecting the entire message.

Furthermore, while attempts have been made to discourage distribution of encoder binaries, Thomson has stated that individuals using free MP3 encoders are not required to pay fees. However, there are a number of disadvantages, which include:. Additionally, patent holders declined to enforce license fees on open source decoders, allowing many free MP3 decoders to develop. The main benefit is the ability to decorate an email with presentational attributes (bold headings etc). In spite of the patent restrictions, the perpetuation of the MP3 format continues; the reasons for this appear to be the network effects caused by:. Use of HTML in email is quite controversial due to a variety of issues. For information about licensing fees see here and here. Many of these clients include a GUI HTML editor for composing emails and a rendering engine for displaying them once received.

Until the key patents expire, open source / free software encoders and players appear to be illegal for commercial use in countries that recognize software patents. Some graphical e-mail clients allow the use of a (often ill-defined) subset of HTML as a pure display language. Microsoft, the makers of the Windows operating system, chose to move away from MP3 to their own proprietary Windows Media formats to avoid the licensing issues associated with the patents. Such behaviour is discouraged due to security problems; even the most notorious offender, Internet Explorer, has mostly abandoned the practice in recent versions (as of 2005). These patent issues significantly slowed the development of unlicensed MP3 software and led to increased focus on creating and popularizing alternatives such as WMA and Ogg Vorbis. Nevertheless, some web browsers do examine the contents or URL of the document and attempt to infer the file type. To make, sell and/or distribute products using the [MPEG Layer-3] standard and thus our patents, you need to obtain a license under these patents from us.". If the MIME type is not recognized as HTML, the web browser should not attempt to render the document as HTML, even if the document is prefaced with a correct Document Type Declaration.

The letter claimed that unlicensed products "infringe the patent rights of Fraunhofer and THOMSON. The exactly same document sent with a HTML MIME type, or served as text/html, might get displayed since the web browser are more lenient with HTML. In September 1998, the Fraunhofer Institute sent a letter to several developers of MP3 software stating that a license was required to "distribute and/or sell decoders and/or encoders". A document sent with an XHTML MIME type, or served as application/xhtml+xml, is expected to be well-formed XML and a syntax error may cause the browser to fail to render the document. See Software patents under the European Patent Convention. In modern browsers, the MIME type that is sent with the HTML document affects how the document is interpreted. Thomson has been granted software patents in EU countries and by the European Patent Office [1], but it is unclear whether or not they would be enforced by courts there. This vital metadata includes the MIME type (text/html for HTML 4.01 and earlier, application/xhtml+xml for XHTML 1.0 and later) and the character encoding (see Character encodings in HTML).

Thomson has been actively enforcing these patents. To allow the web browser to know how to handle the document it received, an indication of the file format of the document must be transmitted along with the document. Thomson Consumer Electronics controls licensing of the MPEG-1/2 Layer 3 patents in countries that recognize software patents, including the United States and Japan, but not EU countries. However, HTTP can be used to serve images, sound and other content in addition to HTML. MP3, which was designed and tuned for use alongside MPEG-1/2 Video, generally performs poorly on monaural data at less than 48 kbit/s or in stereo at less than 80 kbit/s. The World Wide Web is primarily composed of HTML documents transmitted from a web server to a web browser using the HyperText Transfer Protocol (HTTP). Thomson claims that its mp3PRO codec achieves CD quality at 64 kbit/s, but listeners have reported that a 64 kbit/s mp3PRO file compares in quality to a 112 kbit/s MP3 file and does not come reasonably close to CD quality until about 80 kbit/s. See separation of style and content.

The developers of the patent-free Ogg Vorbis codec claim that their algorithm surpasses MP3, RealAudio and WMA sound quality, and the listening tests mentioned above support that claim. CSS provides a way to separate the HTML structure from the content's presentation, by keeping all code dealing with presentation defined in a CSS file. Though proponents of newer codecs such as WMA and RealAudio have asserted that their respective algorithms can achieve CD quality at 64 kbit/s, listening tests have shown otherwise; however, the quality of these codecs at 64 kbit/s is definitely superior to MP3 at the same bitrate. Some of these elements are not permitted in certain varieties of HTML, like HTML 4.01 Strict. What is considered 'CD quality' is quite subjective; for some 128kbit/s MP3 is sufficient, while for others 200kbit/s or higher MP3 is necessary. Standards stress using markup which suggests the structure of the document, like headings, paragraphs, block quoted text, and tables, instead of using markup which is written for visual purposes only, like <font>, <b> (bold), and <i> (italics). At high bitrates (128kbit/s+), most people do not hear significant differences. Efforts of the web development community have led to a new thinking in the way a web document should be written; XHTML epitomizes this effort.

At 64kbit/s, AAC-HE and mp3pro performed marginally better than other codecs. The Transitional DTD was intended to gradually phase in the changes made in the Strict DTD, while the Frameset DTD was intended for those documents which contained frames. At 128kbit/s, Ogg Vorbis, AAC, MPC and WMA Pro tied for first place with LAME MP3 a little behind. In addition to the Strict DTD, HTML 4.01 provides Transitional and Frameset DTDs. Listening tests have attempted to find the best-quality lossy audio codecs at certain bitrates. In some cases, the presence or absence of an appropriate DTD may influence how a web browser will display the page. These include:. This declaration asserts that the document conforms to the Strict DTD of HTML 4.01, which is purely structural, leaving formatting to Cascading Style Sheets.

While they are not similar to MP3, they are good examples of other compression schemes available. For example:. There are also some lossless audio compression methods used on the Internet. In order to specify which version of the HTML standard they conform to, all HTML documents should start with a Document Type Declaration (informally, a "DOCTYPE"), which makes reference to a Document Type Definition (DTD). The Fraunhofer Gesellschaft owns many of the basic patents underlying these codecs, with Dolby Labs, Sony, Thomson Consumer Electronics, and AT&T holding other key patents. Below are the kinds of markup element types in HTML. mp3PRO, MP3, AAC, and MP2 are all members of the same technological family and depend on roughly similar psychoacoustic models. Although perhaps less common now, the shorter form is still widely supported by current software.

Many other lossy audio codecs exist, including:. The most common extension for files containing HTML is .html, however, older operating systems, such as DOS, limit file extensions to three letters, so a .htm extension is also used. Typically, the average volume and clipping information about audio track is stored in the metadata tag. Minor editorial revisions to the HTML 4.0 specification were published as HTML 4.01. The most popular and widely used solution for storing replay gain is known simply as "Replay Gain". HTML 4.0 likewise adopted many browser-specific element types and attributes, but at the same time began to try to "clean up" the standard by marking some of them as deprecated, and suggesting they not be used. The idea is to normalize the volume (not the volume peaks) of audio files, so that the volume does not change between consecutive tracks. Math support as proposed by HTML 3.0 finally came about years later with a different standard, MathML.

A few standards for encoding the gain of an MP3 file have been proposed. HTML 3.1 was never officially proposed, and the next standard proposal was HTML 3.2 (code-named "Wilbur"), which dropped the majority of the new features in HTML 3.0 and instead adopted many browser-specific element types and attributes which had been created for the Netscape and Mosaic web browsers. As compact discs and other various sources are recorded and mastered at different volumes, it is useful to store volume information about a file in the tag so that at playback time, the volume can be dynamically adjusted. Even though it was designed to be compatible with HTML 2.0, it was too complex at the time to be implemented, and when the draft expired in September 1995 work in this direction was discontinued due to lack of browser support. APEv2 can coexist with ID3 tags in the same file, but it can also be used by itself. The HTML 3.0 standard was proposed by the newly formed W3C in March 1995, and provided many new capabilities such as support for tables, text flow around figures, and the display of complex math elements. APEv2 was originally developed for the MPC file format (see the APEv2 specification). Work on HTML+ continued, but it never became a standard.

The most widespread standard tag formats are currently the ID3 ID3v1 and ID3v2 tags, and the more recent APEv2 tag. The first formal specification was therefore given the version number 2.0 in order to distinguish it from these unofficial "standards". A "tag" is data stored in an MP3 (as well as other formats) that contains metadata such as the title, artist, album, track number or other information about the MP3 file to be added to the file itself. Work on a successor for HTML, then called "HTML+", began in late 1993, designed originally to be "A superset of HTML…which will allow a gradual rollover from the previous format of HTML". Therefore, for the most part, comparison of decoders is almost exclusively based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process). That version did not include an IMG element type. The MP3 file has a standard format which is a frame consisting of 384, 576, or 1152 samples (depends on MPEG version and layer) and all the frames have associated header information(32 bits) and side information(9, 17, or 32 bytes, depending on MPEG version and stereo/mono).The header and side information help the decoder to decode the associated huffman encoded data correctly. However, some people consider the initial edition provided by Tim Berners-Lee to be the definitive HTML 1.0.

Most decoders are "bitstream compliant", meaning that the uncompressed output they produce from a given MP3 file will be the same (within a specified degree of rounding tolerance) as the output specified mathematically in the standard document. There is no official standard HTML 1.0 specification because there were multiple informal HTML standards at the time. Decoding, on the other hand, is carefully defined in the standard. . It must be kept in mind that an encoder that is proficient at encoding at higher bitrates (such as LAME, which is in widespread use for encoding at higher bitrates) is not necessarily as good at other, lower bitrates. As such, many consider XHTML to be the "current version" of HTML, but it is a separate, parallel standard; the W3C continues to recommend the use of either XHTML 1.1, XHTML 1.0, or HTML 4.01 for web publishing. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. XHTML, which applies the stricter rules of XML to HTML to make it easier to process and maintain, is the W3C's successor to HTML.

As a result, there are many different MP3 encoders available, each producing files of differing quality. Over time, the trend in the official standards has been to create an increasingly strict language syntax; however, browsers still continue to render pages that are far from valid HTML. This is the domain of psychoacoustics, which aims at understanding how human acoustical perception works (both in our ears and in our brain). Web browsers commonly made assumptions about intent and proceeded with rendering of the page. This is done to limit the temporal spread of quantization noise accompanying the transient. Early versions of HTML were defined with looser syntactic rules which helped its adoption by those unfamiliar with web publishing. If there is a transient 192 samples are taken instead of 576. Later HTML specifications are maintained by the World Wide Web Consortium (W3C).

Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information in the raw audio (or rather its MDCT representation in the frequency domain).During encoding 576 time domain samples are taken and is transformed to 576 frequency domain samples. Originally defined by Tim Berners-Lee and further developed by the IETF with a simplified SGML syntax, HTML is now an international standard (ISO/IEC 15445:2000). The decoding algorithm and file format, as a contrast, are well defined. HTML is used to structure information — denoting certain text as headings, paragraphs, lists and so on — and can be used to describe, to some degree, the appearance and semantics of a document. The MPEG-1 standard does not include a precise specification for an MP3 encoder. In computing, 'HyperText Markup Language ' (HTML) is a markup language designed for the creation of web pages with hypertext and other information to be displayed in a web browser. Nevertheless, a well-tuned MP3 encoder can perform competitively even with these restrictions. potential security issues of simply rendering a complex format like HTML.

In technical terms, MP3 is limited in the following ways:. potential security issues of deluding the recipient to accept an email as being from an authoritative source (such as a bank) when this is not the case; this is related to phishing scams. Newer audio compression formats such as Vorbis and AAC no longer have these limitations. overuse of formatting (there was at one stage a craze for making letterheads using HTML and sending them as part of every e-mail). There are several limitations inherent to the MP3 format that cannot be overcome by using a better encoder. This issue is made slightly worse by the fact that, for compatibility, most clients send a plaintext version as well. Non-standard bitrates up to 640 kbit/s can be achieved with the LAME encoder and the --freeformat option, however only few MP3 players can play those files. the email has larger size because lots of formatting will be much larger than the plain text equivalent.

Some encoders utilize this technique to a great extent. the recipient may not have an email client that can display HTML. This method compares to a sound activated tape recorder that reduces tape consumption by not recording silence. For example,. This technique makes it possible to use more bits for parts of the sound with higher dynamics (more sound movement) and fewer bits for parts with lower dynamics, further increasing quality and decreasing storage space. Links parts of the document to other documents. Audio in MP3 files are divided into frames (which have their own bit rate) so it is possible to change the bit rate dynamically as the file is encoded (although not originally implemented, VBR is in extensive use today). Hypertext markup.

Variable bit rates (VBR) are also possible. For example,. MPEG-2 and [the non-official] MPEG-2.5 includes some additional bit rates: 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s. Describes the appearance of the text, regardless of its function. 44.1 kHz is almost always used (coincides with the sampling rate of compact discs), and 128 kbit/s has become the de facto "good enough" standard, although 192 Kbit/s is becoming increasingly popular over peer-to-peer file sharing networks. Presentational markup. Bit rates available in MPEG-1 Layer 3 are 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, and the available sample frequencies are 32, 44.1 and 48 kHz. For example,.

In the early days of MP3 encoding, a fixed bit rate was used for the entire file. Describes the purpose of text. The general rule is that more information is included from the original sound file when a higher bit rate is used, and thus the higher the quality during play back. Structural markup. The bit rate is variable for MP3 files. (XHTML 2.0, W3C Working Draft). For these operations, the concerns mentioned above are not necessarily relevant, as long as appropriate software (such as mp3DirectCut and MP3Gain) is used to prevent extra decoding-encoding steps. XHTML 1.1, published May 31, 2001.

Some simple editing operations, such as cutting sections of audio, may be performed directly on the encoded MP3 data without necessitating reencoding. XHTML 1.0, published January 26, 2000 as a W3C Recommendation, later revised and republished August 1, 2002. Lossless formats produce the best possible result, at the expense of a lower compression ratio. ISO/IEC 15445:2000 ("ISO HTML", based on HTML 4.01 Strict), published May 15, 2000 as an ISO/IEC international standard. The losses produced by multiple stages of coding may also compound each other, becoming more evident when the signal is reencoded after processing. HTML 4.01, published December 24, 1999 as a W3C Recommendation. Lossless formats are strongly preferred for material that will be edited, mixed, or otherwise processed because the perceptual assumptions made by lossy encoders may not hold true after processing. HTML 4.0, published December 18, 1997 as a W3C Recommendation.

If your aim is to archive sound files with no loss of quality (or work on the sound files in a studio for example), then you should use Lossless compression algorithms, currently capable of compressing 16-bit PCM audio to 38% while leaving the audio identical to the original, such as Lossless Audio LA, Apple Lossless, FLAC, Windows Media Audio 9 Lossless (wma) and Monkey's Audio (among others). HTML 3.2, published January 14, 1997 as a W3C Recommendation. The numbers given above are rough guidelines that work for many people, but in the field of lossy audio compression the only true measure of the quality of a compression process is to listen to the results. HTML 2.0, published November 1995 as IETF RFC 1866, and declared obsolete/historic by RFC 2854 in June 2000. Merely changing the conditions of listening, such as the audio playing system or environment, can expose unwanted distortions caused by lossy compression. Hypertext Markup Language (First Version), published June 1993 as an Internet Engineering Task Force (IETF) working draft (not standard). Individual acoustic perception may vary, so it is not evident that a certain psychoacoustic model can give satisfactory results for everyone.

A given bit rate suffices for some listeners but not for others. It is important to note that quality of an audio signal is subjective. A 128 kbit/s MP3 produced by a good encoder might sound better than a 192 kbit/s MP3 file produced by a bad encoder. It is therefore misleading to speak of 128 kbit/s or 192 kbit/s quality, except in the context of a particular encoder or of the best available encoders.

Good encoders produce acceptable quality at 128 to 160 Kibit/s and near-transparency at 160 to 192 kbit/s, while low quality encoders may never reach transparency, not even at 320 kbit/s. Many early encoders that are no longer widely used:. A few possible encoders:. Different encoders may achieve this with varying degrees of success.

Due to the fact that their lossy encoding loses information, MP3 algorithms work hard to ensure that the parts lost cannot be detected by human listeners by modeling the general characteristics of human hearing (e.g., due to noise masking). Tests may be biased against older formats in favour of new ones by using older encoders based on out-of-date technologies, or even buggy encoders for the old format. When comparing compression schemes, it is important to use encoders that are of equivalent quality. They would contend that more realistic rates would be as follows:.

Many people consider these quoted rates as being heavily skewed in favour of Layer 2 and Layer 3 recordings. However, as different encoders use different models, it is difficult to draw absolute comparisons of this kind. The differences between the layers are caused by the different psychoacoustic models used by them; the Layer 1 algorithm is typically substantially simpler, therefore a higher bit rate is needed for transparent encoding. Fraunhofer Gesellschaft (FhG) publish on their official webpage the following compression ratios and data rates for MPEG-1 Layer 1, 2 and 3, intended for comparison:.

Obviously, imperfections in an MP3 encode will be much less apparent on low-end computer speakers than on a good stereo system connected to a computer or -- especially -- using high-quality headphones. Yet other listeners, and the same listeners in other environments (such as in a noisy moving vehicle or at a party) will consider the quality acceptable. However, listening tests show that with a bit of practice many listeners can reliably distinguish 128 kbit/s MP3s from CD originals; in many cases reaching the point where they consider the MP3 audio to be of unacceptably low quality. For average signals with good encoders, many listeners accept the MP3 bit rate of 128 kibit/s as near enough to compact disc quality for them, providing a compression ratio of approximately 11:1.

As well as the bit rate of the encoded file, the quality of MP3 files depend on the quality of the encoder and the difficulty of the signal being encoded. A good demonstration of compression artifacts is provided by the sound of applause: it is hard to compress because it is random, therefore the failings of the encoder are more obvious, and are audible as ringing. With too low a bit rate, "compression artifacts" (i.e., sounds that were not present in the original recording) may appear in the reproduction. MP3 files encoded with a lower bit rate will generally play back at a lower quality.

By contrast, uncompressed audio as stored on a compact disc has a bit rate of about 1400 kbit/s. Typically rates chosen are between 128 and 256 kilobit per second. Because MP3 is a lossy format, it is able to provide a number of different options for its "bit rate"—that is, the number of bits of encoded data that are used to represent each second of audio. The use of formats that supports DRM is in an attempt to prevent piracy of copyright protected materials, but any computer savvy person can easily rip the DRM from a song file turning it into a file that is not locked to any computer.

Commercial online music distribution services (like the iTunes Music Store) usually prefer other/proprietary music file formats that support Digital Rights Management (DRM) to control and restrict the use of digital music. Due to the vastly increased spread of MP3s through the Internet some major record labels reacted by filing a lawsuit against Napster to protect their Copyrights (see also intellectual property). Controversies regarding peer-to-peer file sharing of MP3 files have flourished in recent years — largely because high compression enables sharing of files that would otherwise be too large and cumbersome to share. Those programs made it very easy for the average user to playback, create, share, and collect MP3s.

MP3 popularity was mostly due to, and interchangeable with, the successes of companies and software packages like Nullsoft's Winamp (released in 1997), mpg123, and Napster (released in 1999). In the first half of 1995 through the late 1990s, MP3 files began flourishing on the Internet. Other founding members include Jon Luini, Brandee Selck, and Ahin Savara. IUMA was started by Rob Lord (who later headed pioneering Nullsoft) and Jeff Patterson, both from the University of California, Santa Cruz, in 1993.

IUMA was the Internet's first high-fidelity music web site, hosting thousands of authorized MP2 recordings before MP3 or the web was popularized. The Internet Underground Music Archive (IUMA) is generally recognized as the start of the on-line music revolution. Initially the only encoder available for MP2 production was the Xing Encoder, accompanied by the program CDDA2WAV, a CD ripper that transformed CD audio tracks to computer data files. In October 1993, MP2 (MPEG-1 Audio Layer 2) files appeared on the Internet and were often played back using the Xing MPEG Audio Player, and later in a program for Unix by Tobias Bading called MAPlay, which was initially released on February 22nd, 1994 (MAPlay was also ported to the Microsoft Windows).

Because of the relatively small hard drives back in that time (~500 MB) the technology was essential to store music for listening pleasure on a computer. With the first real-time software MP3 player Winplay3 (released September 9th, 1995) many people were able to encode and playback MP3 files on their PCs. The filename extension .mp3 was chosen by the Fraunhofer team on July 14, 1995 (previously, the files had been named .bit). Later on, on July 7, 1994 the Fraunhofer Society released the first software MP3 encoder called l3enc.

Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes. Working in non real time on a number of operating systems it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. A reference simulation software written in C language known as ISO 11172-5 was developed by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). Some more serious and critical audio excerpts (glockenspiel, triangle, accordion, ...) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.

This song was chosen because of its softness and simplicity, making it easier to hear imperfections in the compression format during playbacks. Karlheinz Brandenburg used a CD recording of Suzanne Vega's song Tom's Diner to assess the MP3 compression algorithm. Compression ratios with this reference are higher, which demonstrates the problem of the term compression ratio for lossy encoders. Sometimes the Digital Audio Tape (DAT) SP parameters are used (48 kHz, 2x16 bit).

Nevertheless, there are often published compression rates that use the CD parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2x16 bit). Compression efficiency of encoders is typically defined by the bit rate because compression rate depends on the bit depth and sampling rate of the input signal. 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, originally published in 1995. All algorithms were finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3, published in 1993.

Johnston (US), Gerhard Stoll (Germany), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) took ideas from Musicam and ASPEC, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s. Further, on a working group consisting of J.D. Stoll (Layer II). van de Kerkhof (Layer I) and G.

Under the chairmanship of Professor Mussmann (University of Hannover) the editing of the standard was made under the responsibilities of L. Its technologies and ideas were fully incorporated into the definition of ISO MPEG Audio Layer I and Layer II and further on of the Layer III (MP3) format. The Musicam format based on subband coding was key to settle the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame). The Musicam technique, as proposed by Philips (The Netherlands), CCETT (France), IRT (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated to the encoding of high quality compressed audio.

In 1991, there were two proposals available: Musicam (known as Layer 2), and ASPEC (Adaptive Spectral Perceptual Entropy Coding). EU-147 ran from 1987 to 1994. This project was financed by the European Union as a part of the EUREKA research program where it was commonly known as EU-147. MPEG-1 Audio Layer 2 encoding began as the Digital Audio Broadcast (DAB) project managed by Egon Meier-Engelen of the DFVLR (later on called DLR = Deutsche Luft und Raumfahrt = German Aerospace Agency) in Germany.

Nevertheless, any succession is not likely to happen for a significant amount of time due to MP3's overwhelming popularity (MP3 enjoys extremely wide popularity and support, not just by end-users and software but by hardware such as DVD and CD players). In terms of the MPEG specifications, AAC (Advanced audio coding) from MPEG-4 is to be the successor of the MP3 format, although there has been a significant movement to create and popularize other audio formats. MP3 Surround is backward compatible with standard stereo MP3, and file sizes are similar. MP3 Surround, a version of the format supporting 5.1 channels for surround sound, was introduced in December 2004.

The MP3 format uses, at its heart, a hybrid transformation to transform a time domain signal into a frequency domain signal:. MP3 audio can be compressed with different bit rates, providing a range of tradeoffs between data size and sound quality. A number of techniques are employed in MP3 to determine which portions of the audio can be discarded, including psychoacoustics. It provides a representation of pulse-code modulation-encoded (PCM) audio data in a much smaller size by discarding portions that are considered less important to human hearing (similar to JPEG, a lossy compression for images).

MP3 is a compression format. . In popular usage, MP3 also refers to files of sound or music recordings stored in the MP3 format on computers. It was designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners.

MPEG Audio Layer 3, more commonly referred to as MP3, is a popular digital audio encoding and lossy compression format invented and standardized in 1991 by a team of engineers working in the framework of the ISO/IEC MPEG audio committee under the chairmanship of Professor Hans Musmann (University of Hannover - Germany). the majority of home users not knowing or not caring about the software patent controversy, which is in general irrelevant to their choice of the MP3 format for personal use. the lack of DRM-protection technology, which makes MP3 files easy to edit, copy and distribute over networks,. the wide variety of existing software and hardware that takes advantage of the file format,.

the large quantity of music now available in the MP3 format,. the fact that these alternatives do not universally provide a definite advantage over MP3,. familiarity with the format, not knowing alternatives exist,. Apple Lossless.

Wavpack. TTA. SHN, also known as Shorten. Monkey's Audio.

FLAC stands for 'Free Lossless Audio Codec'. Speex, free software and patent free codec based on CELP specifically designed for speech and VoIP. RealAudio from RealNetworks, frequently in use for streaming on websites;. AMR-WB+ Enhanced Adaptive Multi Rate WideBand codec, optimized for cellular and other limited bandwidth use;.

QDesign, used in QuickTime at low bitrates;. Windows Media Audio (WMA) from Microsoft. MPEG-4 AAC, used by Apple's iTunes Music Store and iPod. ATRAC, used in Sony's Minidisc;.

AC-3, used in Dolby Digital and DVD;. mp3PRO from Thomson Multimedia combining MP3 with SBR;. MPC, also known as Musepack (formerly MP+), a derivative of MP2;. Ogg Vorbis from the Xiph.org Foundation, a free software and patent free codec.

MPEG-1/2 Audio Layer 2 (MP2), MP3's predecessor;. Encoder/decoder overall delay is not defined, which means lack of official provision for gapless playback; gaps may be introduced between tracks, although this can be avoided to a degree by using LAME to encode. Joint stereo is done on a frame-to-frame basis. No scale factor band for frequencies above 15.5/15.8 kHz.

Time resolution can be too low for highly transient signals. Bitrate is limited to a maximum of 320 kbit/s. ACM Producer Pro. BladeEnc.

Xing. ISO dist10 reference code. Fraunhofer Gesellschaft: Some encoders are good, some have bugs. It is (in contrast to others) a fully LGPL'd MP3 encoder, with excellent speed and quality, rivaling even MP3's technological successors.

LAME first created by Mike Cheng in early 1998. Layer 3: excellent at 224...320 Kbit/s, very good at 192...224 Kbit/s, good at 128...192 Kbit/s. Layer 2: excellent at 256...384 kbit/s, very good at 224...256 Kbit/s, good at 192...224 Kbit/s. Layer 1: excellent at 384 kbit/s.

Layer 3: 112...128 kbit/s, compression 12:1...10:1. Layer 2: 192...256 kbit/s, compression 8:1...6:1. Layer 1: 384 kbit/s, compression 4:1. Aliasing reduction postprocessing.

36 or 12 tap MDCT; size can be selected independent for sub-band 0...1 and 2...31. 32-band polyphase quadrature filter.