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111-480: AC3 or AC-3 may refer to: Science and technology [ edit ] Dolby AC-3 , Dolby Digital audio codec AC-3 algorithm (Arc Consistency Algorithm 3), one of a series of algorithms used for the solution of constraint satisfaction problems (35414) 1998 AC3 , a minor planet AC-3, an IEC utilization category Ambedkar Community Computing Center (AC3) Transportation [ edit ] Comte AC-3 ,

222-412: A 35 mm release print using sequential data blocks placed between every perforation hole on the soundtrack side of the film. A constant bit rate of 320 kbit/s is used. A charge-coupled device (CCD) scanner in the image projector picks up a scanned video image of this area, and a processor correlates the image area and extracts the digital data as an AC-3 bitstream . The data is then decoded into

333-482: A psychoacoustic coding-algorithm exploiting the masking properties of the human ear. Further optimization by Schroeder and Atal with J.L. Hall was later reported in a 1979 paper. That same year, a psychoacoustic masking codec was also proposed by M. A. Krasner, who published and produced hardware for speech (not usable as music bit-compression), but the publication of his results in a relatively obscure Lincoln Laboratory Technical Report did not immediately influence

444-665: A 1920s Swiss bomber/transport aircraft Southern Pacific class AC-3 , a class of steam locomotive Video games [ edit ] Ace Combat 3 , part of the Ace Combat series of video games Armored Core 3 , part of the Armored Core series of video games Assassin's Creed III , part of the Assassin's Creed series of video games Other [ edit ] Apple Campus 3 , Apple's third large Silicon Valley campus [REDACTED] Topics referred to by

555-486: A 5.1 channel audio source. All film prints with Dolby Digital data also have Dolby Stereo analogue soundtracks using Dolby SR noise reduction and such prints are known as Dolby SR-D prints. The analogue soundtrack provides a fall-back option in case of damage to the data area or failure of the digital decoding; it also provides compatibility with projectors not equipped with digital soundheads. Almost all modern cinema prints are of this type and may also include SDDS data and

666-549: A 5.1-channel 16-bit/48 kHz Dolby Digital format at 640 kbit/s and transports it via a single S/PDIF cable. A similar technology known as DTS Connect is available from competitor DTS . An important benefit of this technology is that it enables the use of digital multichannel sound with consumer sound cards, which are otherwise limited to digital PCM stereo or analog multichannel sound because S/PDIF over RCA, BNC, and TOSLINK can only support two-channel PCM, Dolby Digital multichannel audio, and DTS multichannel audio. HDMI

777-521: A 75–95% reduction in size, depending on the bit rate . In popular usage, MP3 often refers to files of sound or music recordings stored in the MP3 file format (.mp3) on consumer electronic devices. Originally defined in 1991 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

888-460: A bit rate, which specifies how many kilobits per second of audio is desired. The higher the bit rate, the larger the MP3 data stream will be, and, generally, the closer it will sound to the original recording. With too low a bit rate, compression artifacts (i.e., sounds that were not present in the original recording) may be audible in the reproduction. Some audio is hard to compress because of its randomness and sharp attacks. When this type of audio

999-464: A broadcasting system using COFDM modulation was demonstrated on air and in the field 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-chip 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) and

1110-423: A codec called ASPEC, which was submitted to MPEG, and which won the quality competition, but that was mistakenly rejected as too complex to implement. The first practical implementation of an audio perceptual coder (OCF) in hardware (Krasner's hardware was too cumbersome and slow for practical use), was an implementation of a psychoacoustic transform coder based on Motorola 56000 DSP chips. Another predecessor of

1221-483: A core part of the MP3 algorithm. Ernst Terhardt and other collaborators constructed an algorithm describing auditory masking with high accuracy in 1982. This work added to a variety of reports from authors dating back to Fletcher, and to the work that initially determined critical ratios and critical bandwidths. In 1985, Atal and Schroeder presented code-excited linear prediction (CELP), an LPC-based perceptual speech-coding algorithm with auditory masking that achieved

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1332-528: A division of Lucasfilm Ltd. , co-developed Dolby Digital Surround EX ™ for the release of Star Wars: Episode I – The Phantom Menace . Dolby Digital Surround EX has since been used on the DVD releases of the Star Wars prequel and original trilogies. Dolby Digital Live (DDL) is a real-time encoding technology for interactive media such as video games. It converts any audio signals on a PC or game console into

1443-460: A doctoral student at Germany's University of Erlangen-Nuremberg , Karlheinz Brandenburg began working on digital music compression in the early 1980s, focusing on how people perceive music. He completed his doctoral work in 1989. MP3 is directly descended from OCF and PXFM, representing the outcome of the collaboration of Brandenburg — working as a postdoctoral researcher at AT&T-Bell Labs with James D. Johnston ("JJ") of AT&T-Bell Labs — with

1554-570: A full 640 kbit/s. Some Sony PlayStation 2 console games are able to output AC-3 standard audio as well, primarily during pre-rendered cutscenes. Dolby is part of a group of organizations involved in the development of AAC (Advanced Audio Coding), part of MPEG specifications, and considered the successor to MP3. Dolby Digital Plus (DD-Plus) and TrueHD are supported in HD-DVD, as mandatory codecs, and in Blu-ray Disc, as optional codecs. In

1665-539: A given MP3 file will be the same, within a specified degree of rounding tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). Therefore, the comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process). Over time this concern has become less of an issue as CPU clock rates transitioned from MHz to GHz. Encoder/decoder overall delay

1776-488: A home computer as full recordings (as opposed to MIDI notation, or tracker files which combined notation with short recordings of instruments playing single notes). A hacker named SoloH discovered the source code of the "dist10" MPEG reference implementation shortly after the release on the servers of the University of Erlangen . He developed a higher-quality version and spread it on the internet. This code started

1887-483: A large reduction in file sizes when compared to uncompressed audio. The combination of small size and acceptable fidelity led to a boom in the distribution of music over the Internet in the late 1990s, with MP3 serving as an enabling technology at a time when bandwidth and storage were still at a premium. The MP3 format soon became associated with controversies surrounding copyright infringement , music piracy , and

1998-463: A lesser quality setting for lectures and human speech applications and reduces encoding time and complexity. A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3-quality music has risen each year. Berger said the students seem to prefer the 'sizzle' sounds that MP3s bring to music. An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire 's project "The Ghost in

2109-494: A perceptual limitation of human hearing called auditory masking . In 1894, the American physicist Alfred M. Mayer reported that a tone could be rendered inaudible by another tone of lower frequency. In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon. Between 1967 and 1974, Eberhard Zwicker did work in the areas of tuning and masking of critical frequency-bands, which in turn built on

2220-403: A public listening test featuring two early MP3 encoders set at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22. Quality is dependent on the choice of encoder and encoding parameters. This observation caused a revolution in audio encoding. Early on bit rate was the prime and only consideration. At the time MP3 files were of the very simplest type: they used

2331-439: A quarter of MPEG-1 sample rates. For the general field of human speech reproduction, a bandwidth of 5,512 Hz is sufficient to produce excellent results (for voice) using the sampling rate of 11,025 and VBR encoding from 44,100 (standard) WAV file. English speakers average 41–42 kbit/s with -V 9.6 setting but this may vary with the amount of silence recorded or the rate of delivery (wpm). Resampling to 12,000 (6K bandwidth)

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2442-506: 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 kHz sampling rate , a 20 bits/sample input format (the highest available sampling standard in 1991, compatible with the AES/EBU professional digital input studio standard) were

2553-631: A significant data compression ratio for its time. IEEE 's refereed Journal on Selected Areas in Communications reported on a wide variety of (mostly perceptual) audio compression algorithms in 1988. The "Voice Coding for Communications" edition published in February 1988 reported on a wide range of established, working audio bit compression technologies, some of them using auditory masking as part of their fundamental design, and several showing real-time hardware implementations. The genesis of

2664-400: A timecode track to synchronize CD-ROMs carrying DTS soundtracks. The simplest way of converting existing projectors is to add a so-called penthouse digital soundhead above the projector head. However, for new projectors it made sense to use dual analogue/digital soundheads in the normal optical soundhead position under the projector head. To allow for the dual-soundhead arrangement the data

2775-447: Is 1152 samples, divided into two granules of 576 samples. These samples, initially in the time domain, are transformed in one block to 576 frequency-domain samples by MDCT. MP3 also allows the use of shorter blocks in a granule, down to a size of 192 samples; this feature is used when a transient is detected. Doing so limits the temporal spread of quantization noise accompanying the transient (see psychoacoustics ). Frequency resolution

2886-538: Is a coding format for digital audio developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg . 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; for example, compared to CD-quality digital audio , MP3 compression can commonly achieve

2997-414: Is a series of frames; the frame size code is used along with the sample rate code to determine the number of (2-byte) words before the next syncword. Channel blocks can be either long, in which case the entire block is processed as single modified discrete cosine transform , or short, in which case two half length transforms are performed on the block. Below is a simplified AC-3 header. A detailed description

3108-528: Is an advanced lossless audio codec based on Meridian Lossless Packing . Support for the codec was mandatory for HD DVD and is optional for Blu-ray Disc hardware. Dolby TrueHD supports 24-bit bit depths and sample rates up to 192 kHz. Maximum bitrate is 18 Mbit/s while it supports up to 16 audio channels (HD DVD and Blu-ray Disc standards currently limit the maximum number of audio channels to eight). It supports metadata, including dialog normalization and Dynamic Range Control. Although commonly associated with

3219-424: Is compressed, artifacts such as ringing or pre-echo are usually heard. A sample of applause or a triangle instrument with a relatively low bit rate provides good examples of compression artifacts. Most subjective testings of perceptual codecs tend to avoid using these types of sound materials, however, the artifacts generated by percussive sounds are barely perceptible due to the specific temporal masking feature of

3330-546: Is different from Wikidata All article disambiguation pages All disambiguation pages Dolby AC-3 Dolby AC-3 was the original version of the Dolby Digital codec. The basis of the Dolby AC-3 multi-channel audio coding standard is the modified discrete cosine transform (MDCT), a lossy audio compression algorithm. It is a modification of the discrete cosine transform (DCT) algorithm, which

3441-566: Is in the ATSC "Digital Audio Compression (AC-3) (E-AC-3) Standard" , section 5.4. AC3 was covered by patents that expired in March 2017. Patents were used to ask to pay a commercial license to publish an application that decodes AC3. This led some audio app developers to ban AC3 from their apps, although the open source VLC media player supported AC-3 audio without having paid a patent license fee. In Dolby's 2005 original and amended S-1 filings with

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3552-406: Is intended to be mixed with the primary audio soundtrack in the Blu-ray Disc player. Dolby AC-4 is an audio compression standard supporting multiple audio channels and/or audio objects. Support for 5.1 channel audio is mandatory and additional channels up to 7.1.4 are optional. AC-4 provides a 50% reduction in bit rate over AC-3/ Dolby Digital Plus . Dolby TrueHD, developed by Dolby Laboratories,

3663-423: Is limited by the small long block window size, which decreases coding efficiency. Time resolution can be too low for highly transient signals and may cause smearing of percussive sounds. Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. Additionally,

3774-511: Is not backward compatible with existing AC-3 hardware, though E-AC-3 codecs generally are capable of transcoding to AC-3 for equipment connected via S/PDIF . E-AC-3 decoders can also decode AC-3 bitstreams. The fourth generation Apple TV supports E-AC-3. The discontinued HD DVD system directly supported E-AC-3. Blu-ray Disc offers E-AC-3 as an option to graft added channels onto an otherwise 5.1 AC-3 stream, as well as for delivery of secondary audio content (e.g. director's commentary) that

3885-432: Is not defined, which means there is no official provision for gapless playback . However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback. When performing lossy audio encoding, such as creating an MP3 data stream, there is a trade-off between the amount of data generated and the sound quality of the results. The person generating an MP3 selects

3996-884: Is recorded 26 frames ahead of the picture. If a penthouse soundhead is used, the data must be delayed in the processor for the required amount of time, around 2 seconds. This delay can be adjusted in steps of the time between perforations, (approximately 10.4 ms). Dolby Digital remains the predominant sound mixing format for movies, despite the introduction of Dolby Surround 7.1 and Dolby Atmos in 2010 and 2012, respectively. Dolby Digital has similar technologies, included in Dolby Digital EX, Dolby Digital Live, Dolby Digital Plus, Dolby Digital Surround EX, Dolby Digital Recording, Dolby Digital Cinema, Dolby Digital Stereo Creator and Dolby Digital 5.1 Creator. Dolby AC-3 (a backronym for Audio Codec 3, Advanced Codec 3, or Acoustic Coder 3), also known as ATSC A/52 (name of

4107-406: Is that, in any piece of audio, some sections are easier to compress, such as silence or music containing only a few tones, while others will be more difficult to compress. So, the overall quality of the file may be increased by using a lower bit rate for the less complex passages and a higher one for the more complex parts. With some advanced MP3 encoders, it is possible to specify a given quality, and

4218-419: Is the most advanced MP3 encoder. LAME includes a variable bit rate (VBR) encoding which uses a quality parameter rather than a bit rate goal. Later versions (2008+) support an n.nnn quality goal which automatically selects MPEG-2 or MPEG-2.5 sampling rates as appropriate for human speech recordings that need only 5512 Hz bandwidth resolution. In the second half of the 1990s, MP3 files began to spread on

4329-453: Is used on DVD-Video and other purely digital media, like home cinema. In this format, the AC-3 bitstream is interleaved with the video and control bitstreams. The system is used in bandwidth-limited applications other than DVD-Video, such as digital TV. The AC-3 standard allows a maximum coded bit rate of 640 kbit/s. 35 mm film prints use a fixed rate of 320 kbit/s, which is the same as

4440-563: The Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with the use of the term compression ratio for lossy encoders. Karlheinz Brandenburg used a CD recording of Suzanne Vega 's song " Tom's Diner " to assess and refine the MP3 compression algorithm . This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in

4551-594: 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. A reference simulation software implementation, written in the C language and later known as ISO 11172-5 , was developed (in 1991–1996) by the members of the ISO MPEG Audio committee to produce bit-compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). It

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4662-582: The Fraunhofer Institute for Integrated Circuits , Erlangen (where he worked with Bernhard Grill and four other researchers – "The Original Six" ), with relatively minor contributions from the MP2 branch of psychoacoustic sub-band coders. In 1990, Brandenburg became an assistant professor at Erlangen-Nuremberg. While there, he continued to work on music compression with scientists at the Fraunhofer Society 's Heinrich Herz Institute . In 1993, he joined

4773-502: The Institute for Broadcast Technology (Germany), and Matsushita (Japan), was chosen due to its simplicity and error robustness, as well as for its high level of computational efficiency. The MUSICAM format, based on sub-band coding , became the basis for the MPEG Audio compression format, incorporating, for example, its frame structure, header format, sample rates, etc. While much of MUSICAM technology and ideas were incorporated into

4884-599: The Internet , often via underground pirated song networks. The first known experiment in Internet distribution was organized in the early 1990s by the Internet Underground Music Archive , better known by the acronym IUMA. After some experiments using uncompressed audio files, this archive started to deliver on the native worldwide low-speed Internet some compressed MPEG Audio files using the MP2 (Layer II) format and later on used MP3 files when

4995-467: The MPEG-2 ideas and implementation but was named MPEG-2.5 audio since MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered patent holder of MP3, by reducing the frame sync field in the MP3 header from 12 to 11 bits. As in the transition from MPEG-1 to MPEG-2, MPEG-2.5 adds additional sampling rates exactly half of those available using MPEG-2. It thus widens

5106-476: The Nyquist–Shannon sampling theorem . Frequency reproduction is always strictly less than half of the sampling rate, and imperfect filters require a larger margin for error (noise level versus sharpness of filter), so an 8 kHz sampling rate limits the maximum frequency to 4 kHz, while a 48 kHz sampling rate limits an MP3 to a maximum 24 kHz sound reproduction. MPEG-2 uses half and MPEG-2.5 only

5217-465: The Xbox game console and certain nForce2 motherboards, used an early form of this technology. DDL is available on motherboards with codecs such as Realtek 's ALC882D, ALC888DD and ALC888H. Other examples include some C-Media PCI sound cards and Creative Labs' X-Fi and Z series sound cards, whose drivers have enabled support for DDL. NVIDIA later decided to drop DDL support in their motherboards due to

5328-447: The bitstream , called an audio frame, which is made up of 4 parts, the header , error check , audio data , and ancillary data . The MPEG-1 standard does not include a precise specification for an MP3 encoder but does provide examples of psychoacoustic models, rate loops, and the like in the non-normative part of the original standard. MPEG-2 doubles the number of sampling rates that are supported and MPEG-2.5 adds 3 more. When this

5439-509: The subwoofer driven low-frequency effects . Mono and stereo modes are also supported. AC-3 supports audio sample rates up to 48 kHz. In 1991, a limited experimental release of Star Trek VI: The Undiscovered Country in Dolby Digital played in 3 US theatres. In 1992, Batman Returns was the very first movie to be released and presented in Dolby Digital. In 1995, the LaserDisc version of Clear and Present Danger featured

5550-537: The 'Xtreme Audio' and its derivatives such as Prodigy 7.1e, which is incapable of DDL in hardware). X-Fi 's case differs. While they forgot about the plan, programmer Daniel Kawakami made a hot issue by applying Auzentech Prelude DDL module back to Creative X-Fi cards by disguising the hardware identity as Auzentech Prelude. Creative Labs alleged Kawakami violated their intellectual property and demanded he cease distributing his modified drivers. Eventually Creative struck an agreement with Dolby Laboratories regarding

5661-509: The (compressed) audio information in terms of frequencies and amplitudes. The diagram shows that the MP3 Header consists of a sync word , which is used to identify the beginning of a valid frame. This is followed by a bit indicating that this is the MPEG standard and two bits that indicate that layer 3 is used; hence MPEG-1 Audio Layer 3 or MP3. After this, the values will differ, depending on

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5772-442: The 32 sub-band filterbank of Layer II on which the format is based. Besides the bit rate of an encoded piece of audio, the quality of MP3-encoded sound also depends on the quality of the encoder algorithm as well as the complexity of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders do feature quite different quality, even with identical bit rates. As an example, in

5883-579: The 5.1 channel configuration, Dolby Digital allows a number of different channel selections. The options are: These configurations optionally include the extra low-frequency effects (LFE) channel, but only if at least three channels are present. The last two with stereo surrounds can optionally use Dolby Digital EX matrix encoding to add an extra Rear Surround channel, indicated via a 2-bit flag. Many Dolby Digital decoders are equipped with downmixing to distribute encoded channels to speakers. This includes such functions as playing surround information through

5994-418: The 5.1 mix, much like the front center channel on Dolby Pro Logic encoded stereo soundtracks. The result can be played without loss of information on standard 5.1 systems, or played in 6.1 or 7.1 on systems with Surround EX decoding and added speakers. A number of DVDs have a Dolby Digital Surround EX audio option. The theater version of Dolby Digital Surround EX was introduced in 1999, when Dolby and THX ,

6105-504: The DDL pack at no added cost. E-AC-3 (Dolby Digital Plus) is an enhanced coding system based on the AC-3 codec . It offers increased bitrates (up to 6.144 Mbit/s), support for even more audio channels (up to 15.1 discrete channels in the future), and improved coding techniques (only at low data rates) to reduce compression artifacts , enabling lower data rates than those supported by AC-3 (e.g. 5.1-channel audio at 256 kbit/s). It

6216-704: The Dolby license royalty by arranging that the licensing cost be folded into the purchase price of the Creative X-Fi PCI cards rather than as a royalty paid by Creative themselves. Based on the agreement, in September 2008 Creative began selling the Dolby Digital Live packs enabling Dolby Digital Live on Creative's X-Fi PCI series of sound cards. It can be purchased and downloaded from Creative. Subsequently, Creative added their DTS Connect pack to

6327-556: The LaserDisc world AC3RF is the term widely placed on connectors of players that support Dolby Digital. Specific demodulators and receivers from the LaserDisc era (1990s thru early 2000s) also include placement of this term on connectors. LaserDisc titles with Dolby Digital tracks often have the THX logo on their covers. The data layout of AC-3 is described by simplified " C -like" language in official specifications. An AC-3 stream

6438-452: The MP3 file. ISO/IEC 11172-3 defines the range of values for each section of the header along with the specification of the header. Most MP3 files today contain ID3 metadata , which precedes or follows the MP3 frames, as noted in the diagram. The data stream can contain an optional checksum . Joint stereo is done only on a frame-to-frame basis. In short, MP3 compression works by reducing

6549-575: The MP3 format and technology is to be found in the perceptual codec MUSICAM based on an integer arithmetics 32 sub-bands filter bank, driven by a psychoacoustic model. It was primarily designed for Digital Audio Broadcasting (digital radio) and digital TV, and its basic principles were disclosed to the scientific community by CCETT (France) and IRT (Germany) in Atlanta during an IEEE- ICASSP conference in 1991, after having worked on MUSICAM with Matsushita and Philips since 1989. This codec incorporated into

6660-581: The MP3 format. It is nonetheless ubiquitous and especially advantageous for low-bit-rate human speech applications. The ISO standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio) defined three formats: the MPEG-1 Audio Layer I, Layer II and Layer III. The ISO standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Audio) defined an extended version of MPEG-1 Audio: MPEG-2 Audio Layer I, Layer II, and Layer III. MPEG-2 Audio (MPEG-2 Part 3) should not be confused with MPEG-2 AAC (MPEG-2 Part 7 – ISO/IEC 13818-7). LAME

6771-500: The MP3 technology is fully described in a paper from Professor Hans Musmann, who chaired the ISO MPEG Audio group for several years. In December 1988, MPEG called for an audio coding standard. In June 1989, 14 audio coding algorithms were submitted. Because of certain similarities between these coding proposals, they were clustered into four development groups. The first group was ASPEC, by Fraunhofer Gesellschaft , AT&T , France Telecom , Deutsche and Thomson-Brandt . The second group

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6882-402: The MP3" isolates the sounds lost during MP3 compression. In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with

6993-625: The MPEG-1 Audio Layer III standard, MP3 files with a bit rate below 32 kbit/s might be played back sped-up and pitched-up. Earlier systems also lack fast forwarding and rewinding playback controls on MP3. MPEG-1 frames contain the most detail in 320 kbit/s mode, the highest allowable bit rate setting, with silence and simple tones still requiring 32 kbit/s. MPEG-2 frames can capture up to 12 kHz sound reproductions needed up to 160 kbit/s. MP3 files made with MPEG-2 do not have 20 kHz bandwidth because of

7104-642: The SEC, Dolby acknowledged that "Patents relating to our Dolby Digital technologies expire between 2008 and 2017." The last patent covering AC-3 expired March 20, 2017, rendering it free to use. A free ATSC A/52 (AC3) stream decoder, liba52 , is available under the GNU General Public License . FFmpeg and the VLC media player each include code for handling AC-3. MP3 MP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III )

7215-512: The SourceForge website until it became the de facto CBR MP3 encoder. Later an ABR mode was added. Work progressed on true variable bit rate using a quality goal between 0 and 10. Eventually, numbers (such as -V 9.600) could generate excellent quality low bit rate voice encoding at only 41 kbit/s using the MPEG-2.5 extensions. MP3 uses an overlapping MDCT structure. Each MPEG-1 MP3 frame

7326-428: The accuracy of certain components of sound that are considered (by psychoacoustic analysis) to be beyond the hearing capabilities of most humans. This method is commonly referred to as perceptual coding or psychoacoustic modeling. The remaining audio information is then recorded in a space-efficient manner using MDCT and FFT algorithms. The MP3 encoding algorithm is generally split into four parts. Part 1 divides

7437-481: The assessment of music compression codecs. The subband coding technique was found to be efficient, not only for the perceptual coding of high-quality sound materials but especially for the encoding of critical percussive sound materials (drums, triangle ,...), due to the specific temporal masking effect of the MUSICAM sub-band filterbank (this advantage being a specific feature of short transform coding techniques). As

7548-435: The audio signal into smaller pieces, called frames, and an MDCT filter is then performed on the output. Part 2 passes the sample into a 1024-point fast Fourier transform (FFT), then the psychoacoustic model is applied and another MDCT filter is performed on the output. Part 3 quantifies and encodes each sample, known as noise allocation, which adjusts itself to meet the bit rate and sound masking requirements. Part 4 formats

7659-404: The available frequency fidelity in half while likewise cutting the bit rate by 50%. 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. An MP3 coded with MPEG-2 results in half of the bandwidth reproduction of MPEG-1 appropriate for piano and singing. A third generation of "MP3" style data streams (files) extended

7770-431: The combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency. Decoding, on the other hand, is carefully defined in the standard. Most decoders are " bitstream compliant", which means that the decompressed output that they produce from

7881-557: The compression format during playbacks. This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts ( glockenspiel , triangle, accordion , etc.) were taken from

7992-576: The conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference. Bit rate is the product of the sample rate and number of bits per sample used to encode the music. CD audio is 44100 samples per second. The number of bits per sample also depends on the number of audio channels. The CD is stereo and 16 bits per channel. So, multiplying 44100 by 32 gives 1411200—the bit rate of uncompressed CD digital audio. MP3

8103-534: The cost of involved royalties, leaving an empty space in this regard in the sound cards market. Then in June 2005 came Auzentech , which with its X-Mystique PCI card, provided the first consumer sound card with Dolby Digital Live support. Initially no Creative X-Fi based sound cards supported DDL (2005~2007) but a collaboration of Creative and Auzentech resulted in the development of the Auzentech Prelude,

8214-681: The definition of MPEG Audio Layer I and Layer II, the filter bank alone and the data structure based on 1152 samples framing (file format and byte-oriented stream) of MUSICAM remained in the Layer III (MP3) format, as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann of the Leibniz University Hannover , the editing of the standard was delegated to Leon van de Kerkhof (Netherlands), Gerhard Stoll (Germany), and Yves-François Dehery (France), who worked on Layer I and Layer II. ASPEC

8325-597: The encoder will adjust the bit rate accordingly. Users that desire a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate. Perceived quality can be influenced by the listening environment (ambient noise), listener attention, listener training, and in most cases by listener audio equipment (such as sound cards, speakers, and headphones). Furthermore, sufficient quality may be achieved by

8436-475: The file- ripping and sharing services MP3.com and Napster , among others. With the advent of portable media players (including "MP3 players"), a product category also including smartphones , MP3 support remains near-universal and a de facto standard for digital audio. The Moving Picture Experts Group (MPEG) designed MP3 as part of its MPEG-1 , and later MPEG-2 , standards. MPEG-1 Audio (MPEG-1 Part 3), which included MPEG-1 Audio Layer I, II, and III,

8547-664: The first X-Fi card to support DDL. Originally planned to extend DDL support to all X-Fi based sound cards (except the 'Xtreme Audio' line which is incapable of DDL hardware implementation), the plan was dropped because Dolby licensing would have required a royalty payment for all X-Fi cards and, problematically, those already sold. In 2008, Creative released the X-Fi Titanium series of sound cards which fully supports Dolby Digital Live while leaving all PCI versions of Creative X-Fi still lacking support for DDL. Since September 2008, all Creative X-Fi based sound cards support DDL (except

8658-478: The first generation of MP3 defined 14 × 3 = 42 interpretations of MP3 frame data structures and size layouts. The compression efficiency of encoders is typically defined by the bit rate because the compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the compact disc (CD) parameters as references (44.1 kHz , 2 channels at 16 bits per channel or 2×16 bit), or sometimes

8769-486: The front speakers if surround speakers are unavailable, and distributing the center channel to left and right if no center speaker is available. When outputting to separate equipment over a 2-channel connection, a Dolby Digital decoder can optionally encode the output using Dolby Surround to preserve surround information. The '.1' in 5.1, 7.1 etc. refers to the LFE channel, which is also a discrete channel. Dolby Digital audio

8880-485: The fundamental research in the area from Harvey Fletcher and his collaborators at Bell Labs . Perceptual coding was first used for speech coding compression with linear predictive coding (LPC), which has origins in the work of Fumitada Itakura ( Nagoya University ) and Shuzo Saito ( Nippon Telegraph and Telephone ) in 1966. In 1978, Bishnu S. Atal and Manfred R. Schroeder at Bell Labs proposed an LPC speech codec , called adaptive predictive coding , that used

8991-615: The joint stereo coding of MUSICAM and created the MP3 format, which was designed to achieve the same quality at 128  kbit/s as MP2 at 192 kbit/s. The algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991 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. Files or data streams conforming to this standard must handle sample rates of 48k, 44100, and 32k and continue to be supported by current MP3 players and decoders. Thus

9102-482: The main reasons to later adopt the characteristics of MUSICAM as the basic features for an advanced digital music compression codec. During the development of the MUSICAM encoding software, Stoll and Dehery's team made thorough use of a set of high-quality audio assessment material selected by a group of audio professionals from the European Broadcasting Union, and later used as a reference for

9213-513: The mainstream of psychoacoustic codec-development. The discrete cosine transform (DCT), a type of transform coding for lossy compression, proposed by Nasir Ahmed in 1972, was developed by Ahmed with T. Natarajan and K. R. Rao in 1973; they published their results in 1974. This led to the development of the modified discrete cosine transform (MDCT), proposed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987, following earlier work by Princen and Bradley in 1986. The MDCT later became

9324-513: The maximum bit rate for 2-channel MP3 . DVD-Video discs are limited to 448 kbit/s, although many players can successfully play higher-rate bitstreams (which are non-compliant with the DVD specification). HD DVD limits AC-3 to 448 kbit/s. ATSC and digital cable standards limit AC-3 to 448 kbit/s. Blu-ray Disc, the PlayStation 3 and the Xbox game console can output an AC-3 signal at

9435-417: The previous generation for a total of 9 varieties of MP3 format files. The sample rate comparison table between MPEG-1, 2, and 2.5 is given later in the article. MPEG-2.5 is supported by LAME (since 2000), Media Player Classic (MPC), iTunes, and FFmpeg. MPEG-2.5 was not developed by MPEG (see above) and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to

9546-414: The recording industry approved re-incarnation of Napster , and Amazon.com sell unrestricted music in the MP3 format. An MP3 file is made up of MP3 frames, which consist of a header and a data block. This sequence of frames is called an elementary stream . Due to the "bit reservoir", frames are not independent items and cannot usually be extracted on arbitrary frame boundaries. The MP3 Data blocks contain

9657-596: The reproduction of Vega's voice. Accordingly, he dubbed Vega the "Mother of MP3". Instrumental music had been easier to compress, but Vega's voice sounded unnatural in early versions of the format. Brandenburg eventually met Vega and heard Tom's Diner performed live. In 1991, two available proposals were assessed for an MPEG audio standard: MUSICAM ( M asking pattern adapted U niversal S ubband I ntegrated C oding A nd M ultiplexing) and ASPEC ( A daptive S pectral P erceptual E ntropy C oding). The MUSICAM technique, proposed by Philips (Netherlands), CCETT (France),

9768-440: The same bit rate for the entire file: this process is known as constant bit rate (CBR) encoding. Using a constant bit rate makes encoding simpler and less CPU-intensive. However, it is also possible to optimize the size of the file by creating files where the bit rate changes throughout the file. These are known as variable bit rate. The bit reservoir and VBR encoding were part of the original MPEG-1 standard. The concept behind them

9879-448: The same term This disambiguation page lists articles associated with the same title formed as a letter–number combination. If an internal link led you here, you may wish to change the link to point directly to the intended article. Retrieved from " https://en.wikipedia.org/w/index.php?title=AC3&oldid=1221909647 " Category : Letter–number combination disambiguation pages Hidden categories: Short description

9990-866: The sampling rate, MPEG-2 layer III removes all frequencies above half the new sampling rate that may have been present in the source audio. As shown in these two tables, 14 selected bit rates are allowed in MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, along with the 3 highest available sampling rates of 32, 44.1 and 48  kHz . MPEG-2 Audio Layer III also allows 14 somewhat different (and mostly lower) bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling rates of 16, 22.05 and 24  kHz which are exactly half that of MPEG-1. MPEG-2.5 Audio Layer III frames are limited to only 8 bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with 3 even lower sampling rates of 8, 11.025, and 12 kHz. On earlier systems that only support

10101-514: The scope of MP3 to include human speech and other applications yet requires only 25% of the bandwidth (frequency reproduction) possible using MPEG-1 sampling rates. While not an ISO-recognized standard, MPEG-2.5 is widely supported by both inexpensive Chinese and brand-name digital audio players as well as computer software-based MP3 encoders ( LAME ), decoders (FFmpeg) and players (MPC) adding 3 × 8 = 24 additional MP3 frame types. Each generation of MP3 thus supports 3 sampling rates exactly half that of

10212-428: The staff of Fraunhofer HHI. An acapella version of the song " Tom's Diner " by Suzanne Vega was the first song used by Brandenburg to develop the MP3 format. It was used as a benchmark to see how well MP3's compression algorithm handled the human voice. Brandenburg adopted the song for testing purposes, listening to it again and again each time he refined the compression algorithm, making sure it did not adversely affect

10323-401: The standard 5.1 channel Dolby Digital codec in the form of matrixed rear channels, creating 6.1 or 7.1 channel output. It provides an economical and backwards-compatible means for 5.1 soundtracks to carry a sixth, center back surround channel for improved localization of effects. The extra surround channel is matrix encoded onto the discrete left surround and right surround channels of

10434-519: The standard was fully completed. The popularity of MP3s began to rise rapidly with the advent of Nullsoft 's audio player Winamp , released in 1997, which still had in 2023 a community of 80 million active users. In 1998, the first portable solid-state digital audio player MPMan , developed by SaeHan Information Systems, which is headquartered in Seoul , South Korea , was released and the Rio PMP300

10545-447: The standard) or simply Dolby Digital ( DD ), is the common version containing up to six discrete channels of sound. Before 1996 it was marketed as Dolby Surround AC-3 , Dolby Stereo Digital , and Dolby SRD . The most elaborate mode of this codec in common use involves five channels for normal-range speakers ( 20 Hz – 20,000 Hz ) (right, center, left, right surround, left surround) and one channel ( 20 Hz – 120 Hz allotted audio) for

10656-413: The subsequent MPEG-2 standard. MP3 as a file format commonly designates files containing an elementary stream of MPEG-1 Audio or MPEG-2 Audio encoded data, without other complexities of the MP3 standard. Concerning audio compression , which is its most apparent element to end-users, MP3 uses lossy compression to encode data using inexact approximations and the partial discarding of data, allowing for

10767-492: The very first home theater Dolby Digital mix, quickly followed by True Lies , Stargate , Forrest Gump , and Interview with the Vampire among others. Dolby Digital Surround EX (sometimes shortened to Dolby Digital EX) is similar to Dolby's earlier Pro Logic format, which utilized matrix technology to add a center surround channel and single rear surround channel to stereo soundtracks. EX adds an extension to

10878-696: The widespread CD ripping and digital music distribution as MP3 over the internet. 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 42 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 in MPEG-1 Audio. This reduction in sampling rates serves to cut

10989-491: Was MUSICAM , by Matsushita , CCETT , ITT and Philips . The third group was ATAC (ATRAC Coding), by Fujitsu , JVC , NEC and Sony . And the fourth group was SB-ADPCM , by NTT and BTRL. The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF), and Perceptual Transform Coding (PXFM). These two codecs, along with block-switching contributions from Thomson-Brandt, were merged into

11100-485: Was able to demonstrate the first real-time hardware decoding (DSP based) of compressed audio. Some other real-time implementations of MPEG Audio encoders and decoders were available for digital broadcasting (radio DAB , television DVB ) towards consumer receivers and set-top boxes. On 7 July 1994, the Fraunhofer Society released the first software MP3 encoder, called l3enc . The filename extension .mp3

11211-487: Was approved as a committee draft for an ISO / IEC standard in 1991, finalized in 1992, and published in 1993 as ISO/IEC 11172-3:1993. An MPEG-2 Audio (MPEG-2 Part 3) extension with lower sample and bit rates was published in 1995 as ISO/IEC 13818-3:1995. It requires only minimal modifications to existing MPEG-1 decoders (recognition of the MPEG-2 bit in the header and addition of the new lower sample and bit rates). The MP3 lossy compression algorithm takes advantage of

11322-557: Was approved as a committee draft of the ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994. It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on several operating systems, it

11433-460: Was chosen by the Fraunhofer team on 14 July 1995 (previously, the files had been named .bit ). With the first real-time software MP3 player WinPlay3 (released 9 September 1995) many people were able to encode and play back MP3 files on their PCs. Because of the relatively small hard drives of the era (≈500–1000 MB ) lossy compression was essential to store multiple albums' worth of music on

11544-401: Was designed to encode this 1411 kbit/s data at 320 kbit/s or less. If less complex passages are detected by the MP3 algorithms then lower bit rates may be employed. When using MPEG-2 instead of MPEG-1, MP3 supports only lower sampling rates (16,000, 22,050, or 24,000 samples per second) and offers choices of bit rate as low as 8 kbit/s but no higher than 160 kbit/s. By lowering

11655-439: Was later introduced, and it can carry uncompressed multichannel PCM, lossless compressed multichannel audio, and lossy compressed digital audio. However, Dolby Digital Live is still useful with HDMI to allow transport of multichannel audio over HDMI to devices that are unable to handle uncompressed multichannel PCM. Dolby Digital Live is available in sound cards using various manufacturers' audio chipsets. The SoundStorm , used for

11766-590: Was launched in 1999. The ease of creating and sharing MP3s resulted in widespread copyright infringement . Major record companies argued that this free sharing of music reduced sales, and called it " music piracy ". They reacted by pursuing lawsuits against Napster , which was eventually shut down and later sold, and against individual users who engaged in file sharing. Unauthorized MP3 file sharing continues on next-generation peer-to-peer networks . Some authorized services, such as Beatport , Bleep , Juno Records , eMusic , Zune Marketplace , Walmart.com , Rhapsody ,

11877-470: Was proposed by Nasir Ahmed in 1972 for image compression . The DCT was adapted into the MDCT by J.P. Princen, A.W. Johnson and Alan B. Bradley at the University of Surrey in 1987. Dolby Laboratories adapted the MDCT algorithm along with perceptual coding principles to develop the AC-3 audio format for cinema . The AC-3 format was released as the Dolby Digital standard in February 1991. Dolby Digital

11988-460: Was sold afterward in 1998, despite legal suppression efforts by the RIAA . In November 1997, the website mp3.com was offering thousands of MP3s created by independent artists for free. The small size of MP3 files enabled widespread peer-to-peer file sharing of music ripped from CDs, which would have previously been nearly impossible. The first large peer-to-peer filesharing network, Napster ,

12099-500: Was the earliest MDCT-based audio compression standard released, and was followed by others for home and portable usage, such as Sony 's ATRAC (1992), the MP3 standard (1993) and AAC (1997). Batman Returns was the very first movie to be announced as using Dolby SR-D (Spectral Recording-Digital) technology when it premiered in all selected movie theaters in the summer of 1992. Dolby Digital cinema soundtracks are optically recorded on

12210-455: Was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society, and CNET . It provided the highest coding efficiency. A working group consisting of van de Kerkhof, Stoll, Leonardo Chiariglione ( CSELT VP for Media), Yves-François Dehery, Karlheinz Brandenburg (Germany) and James D. Johnston (United States) took ideas from ASPEC, integrated the filter bank from Layer II, added some of their ideas such as

12321-571: Was written, the suggested implementations were quite dated. Implementers of the standard were supposed to devise algorithms suitable for removing parts of the information from the audio input. As a result, many different MP3 encoders became available, each producing files of differing quality. Comparisons were widely available, so it was easy for a prospective user of an encoder to research the best choice. Some encoders that were proficient at encoding at higher bit rates (such as LAME ) were not necessarily as good at lower bit rates. Over time, LAME evolved on

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