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.?YesMP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III) is a for. Originally defined as the third audio format of the standard, it was retained and further extended—defining additional bit-rates and support for more —as the third audio format of the subsequent standard. A third version, known as MPEG 2.5—extended to better support lower bit rates—is commonly implemented, but is not a recognized standard.MP3 (or mp3) as a commonly designates files containing an of MPEG-1 audio and video encoded data, without other complexities of the MP3 standard.In the aspects of MP3 pertaining to —the aspect of the standard most apparent to end-users (and for which is it best known)—MP3 uses to encode data using inexact approximations and the partial discarding of data. This allows 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 mid- to 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, and the file / services and, among others. With the advent of, a product category also including, MP3 support remains near-universal.MP3 compression works by reducing (or approximating) the accuracy of certain components of sound that are considered (by psychoacoustic analysis) to be beyond the of most humans.
This method is commonly referred to as perceptual coding or as modeling. The remaining audio information is then recorded in a space-efficient manner, using and algorithms. Compared to, MP3 compression can commonly achieve a 75 to 95% reduction in size.
For example, an MP3 encoded at a constant bitrate of 128 kbit/s would result in a file approximately 9% of the size of the original CD audio.The (MPEG) designed MP3 as part of its, and later, standards. The first subgroup for audio was formed by several teams of engineers at, and others.
MPEG-1 Audio (MPEG-1 Part 3), which included MPEG-1 Audio Layer I, II and III, was approved as a committee draft for an / standard in 1991, finalised 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). Diagram of the structure of an MP3 file (MPEG version 2.5 not supported, hence 12 instead of 11 bits for MP3 Sync Word).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. Due to the 'byte reservoir', frames are not independent items and cannot usually be extracted on arbitrary frame boundaries. The MP3 Data blocks contain the (compressed) audio information in terms of frequencies and amplitudes. The diagram shows that the MP3 Header consists of a, which is used to identify the beginning of a valid frame. This is followed by a bit indicating that this is the 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 the MP3 file. / 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, which precedes or follows the MP3 frames, as noted in the diagram. The data stream can contain an optional checksum.is done only on a frame-to-frame basis. Encoding and decoding The MP3 encoding algorithm is generally broken into four parts. Part 1 divides the audio signal into smaller pieces, called frames, and a (MDCT) filter is then performed on the output. Part 2 passes the sample into a 1024-point (FFT), then the 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 in order to meet the and requirements. Part 4 formats the, called an audio frame, which is made up of 4 parts, the, and.The standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard.MPEG-2 doubles the number of sampling rates which are supported and MPEG-2.5 adds 3 more.
When this was written, the suggested implementations were quite dated. Implementers of the standard were supposed to devise their own 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 ) were not necessarily as good at lower bit rates.
Over time, LAME evolved on 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.During encoding, 576 time-domain samples are taken and are transformed to 576. If there is a, 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient.
(See.) Frequency resolution 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, 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 are ' compliant', which means that the decompressed output that they produce from a given MP3 file will be the same, within a specified degree of tolerance, as the output specified mathematically in the ISO/IEC high standard document (ISO/IEC 11172-3). Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much or time they use in the decoding process). Over time this concern has become less of an issue as CPU speeds transitioned from MHz to GHz. Encoder/decoder overall delay is not defined, which means there is no official provision for. However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback.Quality 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 a, which specifies how many 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, (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 is compressed, artifacts such as ringing or are usually heard. A sample of applause or a triangle instrument with a relatively low bit rate provide 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 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 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 bitrate was the prime and only consideration. At the time MP3 files were of the very simplest type: they used the same bit rate for the entire file: this process is known as (CBR) encoding.
Using a constant bit rate makes encoding simpler and less CPU intensive. However, it is also possible to create files where the bit rate changes throughout the file. These are known as. The bit reservoir and VBR encoding were actually part of the original MPEG-1 standard. The concept behind them 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 the encoder will adjust the bit rate accordingly. Users that desire a particular 'quality setting' that is 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 listening environment (ambient noise), listener attention, and listener training and in most cases by listener audio equipment (such as sound cards, speakers and headphones). Furthermore, sufficient quality may be achieved by a lesser quality setting for lectures and human speech applications and reduces encoding time and complexity. A test given to new students by 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 's project 'The Ghost in 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 the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference. Bit rate MPEG Audio Layer IIIavailable bit rates (kbit/s) MPEG-1Audio Layer IIIMPEG-2Audio Layer IIIMPEG-2.5Audio Layer III–88–160–9696–128–n/a144–160160–192––224––256––320––Supported sampling ratesby MPEG Audio Format MPEG-1Audio Layer IIIMPEG-2Audio Layer IIIMPEG-2.5Audio Layer III––8000 Hz––11025 Hz––12000 Hz–16000 Hz––22050 Hz––24000 Hz–32000 Hz––44100 Hz––48000 Hz––Bitrate 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. CD is stereo and 16 bits per channel. So, multiplying 44100 by 32 gives 1411200—the bitrate of uncompressed CD digital audio. MP3 was designed to encode this 1411 kbit/s data at 320 kbit/s or less. As less complex passages are detected by MP3 algorithms then lower bitrates may be employed.
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When using MPEG-2 instead of MPEG-1, MP3 supports only lower sampling rates (16000, 22050 or 24000 samples per second) and offers choices of bitrate as low as 8 kbit/s but no higher than 160 kbit/s. By lowering 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 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 of 32, 44.1 and 48. MPEG-2 Audio Layer III also allows 14 somewhat different (and mostly lower) of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with of 16, 22.05 and 24 which are exactly half that of MPEG-1 MPEG-2.5 Audio Layer III frames are limited to only 8 of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with 3 even lower of 8, 11.025, and 12 kHz. MPEG-1 frames contain the most detail in 320 kbit/s mode 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 don't have 20 kHz bandwidth because of the.
Frequency reproduction is always strictly less than half of the sampling frequency, 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 a quarter of MPEG-1 sample rates.For the general field of human speech reproduction, a bandwidth of 5512 Hz is sufficient to produce excellent results (for voice) using the sampling rate of 11025 and VBR encoding from 44100 (standard) WAV file. This is easily accomplished using LAME version 3.99.5 and the command line 'lame -V 9.6 lecture.WAV' English speakers average 41–42 kbit/s with -V 9.6 setting but this may vary with amount of silence recorded or the rate of delivery (wpm). Resampling to 12000 (6K bandwidth) is selected by the LAME parameter -V 9.4 Likewise -V 9.2 selects 16000 sample rate and a resultant 8K lowpass filtering.
For more info see Nyquist – Shannon. Older versions of LAME and FFmpeg only support integer arguments for variable bit rate quality selection parameter. The n.nnn quality parameter (-V) is documented at lame.sourceforge.net but is only supported in LAME with the new style VBR variable bit rate quality selector—not average bit rate (ABR).A sample rate of 44.1 kHz is commonly used for music reproduction, because this is also used for, the main source used for creating MP3 files.
A great variety of bit rates are used on the Internet. A bit rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread. Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent of approximately 11:1, 9:1 and 7:1 respectively.Non-standard bit rates up to 640 kbit/s can be achieved with the encoder and the freeformat option, although few MP3 players can play those files.
According to the ISO standard, decoders are only required to be able to decode streams up to 320 kbit/s. Early MPEG Layer III encoders used what is now called (CBR). The software was only able to use a uniform bitrate on all frames in an MP3 file. Later more sophisticated MP3 encoders were able to use the bit reservoir to target an selecting the encoding rate for each frame based on the complexity of the sound in that portion of the recording.A more sophisticated MP3 encoder can produce audio. MPEG audio may use bitrate switching on a per-frame basis, but only layer III decoders must support it. VBR is used when the goal is to achieve a fixed level of quality. The final file size of a VBR encoding is less predictable than with.
Is a type of VBR implemented as a compromise between the two: the bitrate is allowed to vary for more consistent quality, but is controlled to remain near an average value chosen by the user, for predictable file sizes. Although an MP3 decoder must support VBR to be standards compliant, historically some decoders have bugs with VBR decoding, particularly before VBR encoders became widespread. The most evolved LAME MP3 encoder supports the generation of VBR, ABR, and even the ancient CBR MP3 formats.Layer III audio can also use a 'bit reservoir', a partially full frame's ability to hold part of the next frame's audio data, allowing temporary changes in effective bitrate, even in a constant bitrate stream. Internal handling of the bit reservoir increases encoding delay.
There is no scale factor band 21 (sfb21) for frequencies above approx 16, forcing the encoder to choose between less accurate representation in band 21 or less efficient storage in all bands below band 21, the latter resulting in wasted bitrate in VBR encoding. Ancillary data The ancillary data field can be used to store user defined data. The ancillary data is optional and the number of bits available is not explicitly given. The ancillary data is located after the Huffman code bits and ranges to where the next frame's maindatabegin points to. Encoder used ancillary data to encode extra information which could improve audio quality when decoded with its own algorithm.Metadata. Main articles: andA 'tag' in an audio file is a section of the file that contains such as the title, artist, album, track number or other information about the file's contents. The MP3 standards do not define tag formats for MP3 files, nor is there a standard that would support metadata and obviate the need for tags.
However, several de facto standards for tag formats exist. As of 2010, the most widespread are, and the more recently introduced.
These tags are normally embedded at the beginning or end of MP3 files, separate from the actual MP3 frame data. MP3 decoders either extract information from the tags, or just treat them as ignorable, non-MP3 junk data.Playing & editing software often contains tag editing functionality, but there are also applications dedicated to the purpose.
Aside from metadata pertaining to the audio content, tags may also be used for. Is a standard for measuring and storing the loudness of an MP3 file in its metadata tag, enabling a ReplayGain-compliant player to automatically adjust the overall playback volume for each file.
May be used to reversibly modify files based on ReplayGain measurements so that adjusted playback can be achieved on players without ReplayGain capability.Licensing, ownership and legislation The basic MP3 decoding and encoding technology is patent-free in the European Union, all patents having expired there by 2012 at the latest. In the United States, the technology became substantially patent-free on 16 April 2017 (see below). MP3 patents expired in the US between 2007 and 2017. In the past, many organizations have claimed ownership of related to MP3 decoding or encoding. These claims led to a number of legal threats and actions from a variety of sources. As a result, uncertainty about which patents must be licensed in order to create MP3 products without committing patent infringement in countries that allow was a common feature of the early stages of adoption of the technology.The initial near-complete MPEG-1 standard (parts 1, 2 and 3) was publicly available on 6 December 1991 as ISO CD 11172. In most countries, patents cannot be filed after prior art has been made public, and patents expire 20 years after the initial filing date, which can be up to 12 months later for filings in other countries.
As a result, patents required to implement MP3 expired in most countries by December 2012, 21 years after the publication of ISO CD 11172.An exception is the United States, where patents in force but filed prior to 8 June 1995 expire after the later of 17 years from the issue date or 20 years from the priority date. A lengthy patent prosecution process may result in a patent issuing much later than normally expected (see ). The various MP3-related patents expired on dates ranging from 2007 to 2017 in the United States. Patents for anything disclosed in ISO CD 11172 filed a year or more after its publication are questionable. If only the known MP3 patents filed by December 1992 are considered, then MP3 decoding has been patent-free in the US since 22 September 2015, when, which had a PCT filing in October 1992, expired.
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If the longest-running patent mentioned in the aforementioned references is taken as a measure, then the MP3 technology became patent-free in the United States on 16 April 2017, when, held and administered by, expired. As a result, many projects, such as the, have decided to start shipping MP3 support by default, and users will no longer have to resort to installing unofficial packages maintained by third party software repositories for MP3 playback or encoding.(formerly called Thomson Consumer Electronics) claimed to control MP3 licensing of the Layer 3 patents in many countries, including the United States, Japan, Canada and EU countries. Technicolor had been actively enforcing these patents. MP3 license revenues from Technicolor's administration generated about €100 million for the Fraunhofer Society in 2005. 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'.
The letter claimed that unlicensed products 'infringe the patent rights of Fraunhofer and Thomson. To make, sell or distribute products using the MPEG Layer-3 standard and thus our patents, you need to obtain a license under these patents from us.' This led to the situation where the MP3 encoder project could not offer its users official binaries that could run on their computer.
The project's position was that as source code, LAME was simply a description of how an MP3 encoder could be implemented. Unofficially, compiled binaries were available from other sources.Sisvel S.p.A. And its United States subsidiary Audio MPEG, Inc.
Previously sued Thomson for patent infringement on MP3 technology, but those disputes were resolved in November 2005 with Sisvel granting Thomson a license to their patents. Motorola followed soon after, and signed with Sisvel to license MP3-related patents in December 2005. Except for three patents, the US patents administered by Sisvel had all expired in 2015.
The three exceptions are:, expired February 2017;, expired February 2017; and, expired 9 April 2017.In September 2006, German officials seized MP3 players from 's booth at the in Berlin after an Italian patents firm won an injunction on behalf of Sisvel against SanDisk in a dispute over licensing rights. The injunction was later reversed by a Berlin judge, but that reversal was in turn blocked the same day by another judge from the same court, 'bringing the Patent Wild West to Germany' in the words of one commentator. In February 2007, Texas MP3 Technologies sued Apple, Samsung Electronics and Sandisk in, claiming infringement of a portable MP3 player patent that Texas MP3 said it had been assigned. Apple, Samsung, and Sandisk all settled the claims against them in January 2009.has asserted several MP3 coding and compression patents, allegedly inherited from AT&T-Bell Labs, in litigation of its own. In November 2006, before the companies' merger, Alcatel for allegedly infringing seven patents. On 23 February 2007, a San Diego jury awarded US $1.52 billion in damages for infringement of two of them. The court subsequently revoked the award, however, finding that one patent had not been infringed and that the other was not owned by; it was co-owned by and Fraunhofer, who had licensed it to, the judge ruled.
That defense judgment was upheld on appeal in 2008. See for more information.Alternative technologies. Main article:Other lossy formats exist. Among these, (AAC) is the most widely used, and was designed to be the successor to MP3.
There also exist other lossy formats such as. They are members of the same technological family as MP3 and depend on roughly similar and algorithms. Whereas MP3 uses a hybrid coding approach that is part MDCT and part, AAC is purely MDCT, significantly improving compression efficiency. Many of the basic underlying these formats are held by, Alcatel-Lucent, and.There are also open compression formats like and that are available free of charge and without any known patent restrictions. Some of the newer audio compression formats, such as AAC, WMA Pro and Vorbis, are free of some limitations inherent to the MP3 format that cannot be overcome by any MP3 encoder.Besides lossy compression methods, are a significant alternative to MP3 because they provide unaltered audio content, though with an increased file size compared to lossy compression.
Lossless formats include (Free Lossless Audio Codec), and many others.See also.References.