MP3

Source: Wikipedia, the free encyclopedia.
For other uses, see MP3 (disambiguation).
Not to be confused with MPEG-3.

MP3
Filename extension.mp3
.bit (before 1995)[1]
Internet media type
  • audio/mpeg[2]
  • audio/MPA[3]
  • audio/mpa-robust[4]
Developed by
ISO/IEC 13818-3[8]
Open format?Yes[9]
Free format?Expired patents[10]

MP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III)[4] is a coding format for digital audio developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg,[11][12] with support from other digital scientists in other countries. Originally defined 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 the subsequent MPEG-2 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

elementary stream
of MPEG-1 Audio or MPEG-2 Audio encoded data, without other complexities of the MP3 standard.

Concerning

smartphones
, 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 hearing capabilities of most humans. This method is commonly referred to as perceptual coding or psychoacoustic modeling.[13] The remaining audio information is then recorded in a space-efficient manner using MDCT and FFT algorithms. Compared to CD-quality digital audio, MP3 compression can commonly achieve a 75–95% reduction in size. For example, an MP3 encoded at a constant bit rate of 128 kbit/s would result in a file approximately 9% the size of the original CD audio.[14] In the early 2000s, compact disc players increasingly adopted support for playback of MP3 files on data CDs.

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, was approved as a committee draft for an ISO/IEC standard in 1991,[15][16] finalized in 1992,[17] and published in 1993 as ISO/IEC 11172-3:1993.[7] 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.[8][18] 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).

History

Background

Further information: Linear predictive coding and Modified discrete cosine transform

The MP3 lossy

audio-data compression algorithm takes advantage of 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.[19] In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon.[20] Between 1967 and 1974, Eberhard Zwicker did work in the areas of tuning and masking of critical frequency-bands,[21][22] which in turn built on the fundamental research in the area from Harvey Fletcher and his collaborators at Bell Labs.[23]

Perceptual coding was first used for

Lincoln Laboratory Technical Report[29]
did not immediately influence the mainstream of psychoacoustic codec-development.

The

Nasir Ahmed in 1972, was developed by Ahmed with T. Natarajan and K. R. Rao in 1973; they published their results in 1974.[30][31][32] 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,[33] following earlier work by Princen and Bradley in 1986.[34] The MDCT later became a core part of the MP3 algorithm.[35]

Ernst Terhardt and other collaborators constructed an algorithm describing auditory masking with high accuracy in 1982.[36] 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

IEEE's refereed Journal on Selected Areas in Communications reported on a wide variety of (mostly perceptual) audio compression algorithms in 1988.[37] The "Voice Coding for Communications" edition published in February 1988 reported on a wide range of established, working audio bit compression technologies,[37]
some of them using auditory masking as part of their fundamental design, and several showing real-time hardware implementations.

Development

The genesis of the MP3 technology is fully described in a paper from Professor Hans Musmann,

SB-ADPCM, by NTT and BTRL.[38]

The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF),

DSP
chips.

Another predecessor of 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,[41] after having worked on MUSICAM with Matsushita and Philips since 1989.[38]

This codec incorporated into a broadcasting system using COFDM modulation was demonstrated on air and in the field

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 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[43] selected by a group of audio professionals from the European Broadcasting Union, and later used as a reference for 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 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.[44] 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 the Fraunhofer Institute for Integrated Circuits, Erlangen (where he worked with Bernhard Grill and four other researchers – "The Original Six"[45]), 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 the staff of Fraunhofer HHI.[44] 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 the reproduction of Vega's voice.[46] Accordingly, he dubbed Vega the "Mother of MP3".[47]
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.

Standardization

In 1991, two available proposals were assessed for an MPEG audio standard:

MUSICAM (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The MUSICAM technique, proposed by Philips (Netherlands), CCETT (France), the Institute for Broadcast Technology (Germany), and Matsushita (Japan),[48] was chosen due to its simplicity and error robustness, as well as for its high level of computational efficiency.[49] 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 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 was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society, and CNET.[50] It provided the highest coding efficiency.

A

kbit/s as MP2
at 192 kbit/s.

The algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991

MP3 players
and decoders. Thus 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

kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes 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

compression algorithm.[51] This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in 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 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.[citation needed
]

Going public

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 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.

DVB
) towards consumer receivers and set-top boxes.

On 7 July 1994, the Fraunhofer Society released the first software MP3 encoder, called

tracker
files which combined notation with short recordings of instruments playing single notes).

Fraunhofer example implementation

A hacker named SoloH discovered the

University of Erlangen. He developed a higher-quality version and spread it on the internet. This code started the widespread CD ripping and digital music distribution as MP3 over the internet.[58][59][60][61]

Further versions

Further work on MPEG audio[62] 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[18]), originally published in 1995.[8][63] 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 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.[62] 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 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 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 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.[64][65] 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 the MP3 format. It is nonetheless ubiquitous and especially advantageous for low-bit-rate human speech applications.

MPEG Audio Layer III versions
Version International Standard[*] First edition public release date Latest edition public release date
MPEG-1 Audio Layer III ISO/IEC 11172-3 (MPEG-1 Part 3)[7][16] 1993
MPEG-2 Audio Layer III ISO/IEC 13818-3 (MPEG-2 Part 3)[8][66] 1995 1998
MPEG-2.5 Audio Layer III nonstandard, Fraunhofer proprietary[64][65] 2000 2008

* 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).[18]

LAME is the most advanced MP3 encoder.[

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.

Internet distribution

In the second half of the 1990s, MP3 files began to spread on the

RIAA.[69]

In November 1997, the website

mp3.com was offering thousands of MP3s created by independent artists for free.[69] 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, 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.[70]

Unauthorized MP3 file sharing continues on next-generation

Amazon.com
sell unrestricted music in the MP3 format.

Design

File structure

Diagram of the structure of an MP3 file
Diagram of the structure of an MP3 file (MPEG version 2.5, not described here, changes the last bit of sync word to "0" as an indication, effectively moving one bit to the version field[65]).

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

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 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.[71]

Encoding and decoding

The MP3 encoding algorithm is generally split into four parts. Part 1 divides 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

audio data, and ancillary data.[35]

The

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 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.[73] 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 is limited by the small long block window size, which decreases coding efficiency.[71] Time resolution can be too low for highly transient signals and may cause smearing of percussive sounds.[71]

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.[71] 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.[74]

Decoding, on the other hand, is carefully defined in the standard. Most

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 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.

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 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 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 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,[75] 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.[76]

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 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 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 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 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.[77]

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire'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",[78][79][80] 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.[81]

Bit rate

MPEG Audio Layer III
available bit rates (kbit/s)[16][64][65][66][82]
MPEG-1
Audio Layer III 		MPEG-2
Audio Layer III 		MPEG-2.5
Audio Layer III
– 8 8
– 16 16
– 24 24
32 32 32
40 40 40
48 48 48
56 56 56
64 64 64
80 80 –
96 96 –
112 112 –
128 128 –
n/a 144 –
160 160 –
192 – –
224 – –
256 – –
320 – –
Supported sampling rates
by MPEG Audio Format[16][64][65][66]
MPEG-1
Audio Layer III 		MPEG-2
Audio Layer III 		MPEG-2.5
Audio Layer III
– – 8 kHz
– – 11.025 kHz
– – 12 kHz
– 16 kHz –
– 22.05 kHz –
– 24 kHz –
32 kHz – –
44.1 kHz – –
48 kHz – –

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 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 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 which are exactly half that of MPEG-1.[65] 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.[citation needed
] On earlier systems that only support 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.[83][84]

MPEG-1 frames contain the most detail in 320 kbit/s mode, the highest allowable bit rate setting,[85] 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 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 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) is selected by the LAME parameter -V 9.4. Likewise -V 9.2 selects a 16,000 sample rate and a resultant 8K lowpass filtering. Older versions of LAME and FFmpeg only support integer arguments for the 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

CD audio, 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,[86] at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth 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 × 44,100 samples/second × 2 channels / 1,000 bits/kilobit), so the bit rates 128, 160, and 192 kbit/s represent compression ratios
of approximately 11:1, 9:1 and 7:1 respectively.

Non-standard bit rates up to 640 kbit/s can be achieved with the

average bit rate
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 variable bit rate audio. MPEG audio may use bit rate switching on a per-frame basis, but only layer III decoders must support it.[65][90][91][92] 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 constant bit rate. Average bit rate is a type of VBR implemented as a compromise between the two: the bit rate 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 older 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 bit rate, even in a constant bit rate stream.

kHz, 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 bit rate in VBR encoding.[93]

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 main_data_begin points to. Encoder mp3PRO used ancillary data to encode extra information which could improve audio quality when decoded with its algorithm.

Metadata

Main articles:
APEv2 tag

A "tag" in an audio file is a section of the file that contains

APEv2
. 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 and editing software often contains tag editing functionality, but there are also tag editor applications dedicated to the purpose. Aside from metadata about the audio content, tags may also be used for DRM.[94] ReplayGain is a standard for measuring and storing the loudness of an MP3 file (audio normalization) in its metadata tag, enabling a ReplayGain-compliant player to automatically adjust the overall playback volume for each file. MP3Gain 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 patents related to MP3 decoding or encoding. These claims led to several legal threats and actions from a variety of sources. As a result, in countries that allow software patents, uncertainty about which patents must have been licensed to create MP3 products without committing patent infringement was common in the early stages of the technology's adoption.

The initial near-complete MPEG-1 standard (parts 1, 2, and 3) was publicly available on 6 December 1991 as ISO CD 11172.[95][96] 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 before 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 issued much later than normally expected (see

Fedora operating system, 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.[103]

Technicolor (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.[104] Technicolor had been actively enforcing these patents.[105] MP3 license revenues from Technicolor's administration generated about €100 million for the Fraunhofer Society in 2005.[106] 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."[107] This led to the situation where the LAME
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., a Luxembourg-based company, administers licenses for patents applying to MPEG Audio.[108] They, along with its United States subsidiary Audio MPEG, Inc. previously sued Thomson for patent infringement on MP3 technology,[109] 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.[110] Except for three patents, the US patents administered by Sisvel[111] had all expired in 2015. The three exceptions are: U.S. patent 5,878,080, expired February 2017; U.S. patent 5,850,456, expired February 2017; and U.S. patent 5,960,037, expired 9 April 2017. As of around the first quarter of 2023, Sisvel's licensing program has become a legacy.[112]

In September 2006, German officials seized MP3 players from

IFA show 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,[113] 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.[114] In February 2007, Texas MP3 Technologies sued Apple, Samsung Electronics and Sandisk in eastern Texas federal court, 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.[115][116]

AT&T and Fraunhofer, who had licensed it to Microsoft, the judge ruled.[118] That defense judgment was upheld on appeal in 2008.[119]

Alternative technologies

Main article: List of codecs

Other lossy formats exist. Among these,

JVC Kenwood, Philips, Microsoft, and NTT.[122]

When the digital audio player market was taking off, MP3 was widely adopted as the standard hence the popular name "MP3 player". Sony was an exception and used their own ATRAC codec taken from their MiniDisc format, which Sony claimed was better.[123] Following criticism and lower than expected Walkman sales, in 2004 Sony for the first time introduced native MP3 support to its Walkman players.[124]

There are also open compression formats like Opus and Vorbis that are available free of charge and without any known patent restrictions. Some of the newer audio compression formats, such as AAC, WMA Pro, Vorbis, and Opus, are free of some limitations inherent to the MP3 format that cannot be overcome by any MP3 encoder.[97][125]

Besides lossy compression methods,

Apple Lossless
and many others.

See also

References

  1. ^ a b "Happy Birthday MP3!". Fraunhofer IIS. 12 July 2005. Retrieved 18 July 2010.
  2. doi:10.17487/RFC3003. Retrieved 7 December 2009. {{cite journal}}: Cite journal requires |journal= (help
    )
  3. doi:10.17487/RFC3555. Retrieved 7 December 2009. {{cite journal}}: Cite journal requires |journal= (help
    )
  4. ^
    doi:10.17487/RFC5219. Retrieved 4 December 2014. {{cite journal}}: Cite journal requires |journal= (help
    )
  5. ^ "The mp3 team". Fraunhofer IIS. Retrieved 12 June 2020.
  6. S2CID 3773268
    . Reference 3 in the paper is to Committee Draft of Standard ISO/IEC 11172, December 6, 1991.
  7. ^ a b c d "ISO/IEC 11172-3:1993 – Information technology — Coding of moving pictures and associated audio for digital storage media at up to about 1,5 Mbit/s — Part 3: Audio". ISO. 1993. Retrieved 14 July 2010.
  8. ^ a b c d "ISO/IEC 13818-3:1995 – Information technology — Generic coding of moving pictures and associated audio information — Part 3: Audio". ISO. 1995. Retrieved 14 July 2010.
  9. ^ "MP3 technology at Fraunhofer IIS". Fraunhofer IIS. Archived from the original on 15 August 2021. Retrieved 12 June 2020.
  10. ^ MP3 (MPEG Layer III Audio Encoding) (Full draft). Sustainability of Digital Formats. Washington, D.C.: Library of Congress. 3 May 2017. Retrieved 1 December 2021.
  11. ^ "73. "Father" of the MP3, Karlheinz Brandenburg" – via www.youtube.com.
  12. ^ "On the 20th Birthday of the MP3, An Interview With The "Father" of the MP3, Karlheinz Brandenburg".
  13. doi:10.1109/5.241504
    .
  14. ^ "MP3 (MPEG Layer III Audio Encoding)". The Library of Congress. 27 July 2017. Retrieved 9 November 2017.
  15. ^ a b ISO (November 1991). "MPEG Press Release, Kurihama, November 1991". ISO. Archived from the original on 3 May 2011. Retrieved 17 July 2010.
  16. ^ a b c d e ISO (November 1991). "CD 11172-3 – CODING OF MOVING PICTURES AND ASSOCIATED AUDIO FOR DIGITAL STORAGE MEDIA AT UP TO ABOUT 1.5 MBIT/s Part 3 AUDIO" (PDF). Archived from the original (PDF) on 30 December 2013. Retrieved 17 July 2010.
  17. ^ a b ISO (6 November 1992). "MPEG Press Release, London, 6 November 1992". Chiariglione. Archived from the original on 12 August 2010. Retrieved 17 July 2010.
  18. ^ a b c ISO (October 1998). "MPEG Audio FAQ Version 9 – MPEG-1 and MPEG-2 BC". ISO. Retrieved 28 October 2009.
  19. doi:10.1080/14786449408620544
    .
  20. doi:10.1121/1.1907853
    .
  21. ISBN 978-3-642-65904-1
    .
  22. ^ Zwicker, Eberhard; Feldtkeller, Richard (1999) [1967]. Das Ohr als Nachrichtenempfänger [The Ear as a Communication Receiver]. Trans. by Hannes Müsch, Søren Buus, and Mary Florentine. Archived from the original on 14 September 2000. Retrieved 29 June 2008.
  23. ISBN 978-1-56396-393-3
    .
  24. ^
    ISBN 978-3-319-05660-9
    .
  25. ISSN 1932-8346
    .
  26. doi:10.1109/ICASSP.1978.1170564
    .
  27. doi:10.1121/1.383662
    .
  28. hdl:1721.1/16011
    .
  29. ^ Krasner, M. A. (18 June 1979). "Digital Encoding of Speech Based on the Perceptual Requirement of the Auditory System (Technical Report 535)" (PDF). Archived (PDF) from the original on 3 September 2017.
  30. doi:10.1016/1051-2004(91)90086-Z
    .
  31. S2CID 149806273
  32. ISBN 978-0-12-580203-1
  33. ^ J. P. Princen, A. W. Johnson und A. B. Bradley: Subband/transform coding using filter bank designs based on time domain aliasing cancellation, IEEE Proc. Intl. Conference on Acoustics, Speech, and Signal Processing (ICASSP), 2161–2164, 1987
  34. ^ John P. Princen, Alan B. Bradley: Analysis/synthesis filter bank design based on time domain aliasing cancellation, IEEE Trans. Acoust. Speech Signal Processing, ASSP-34 (5), 1153–1161, 1986
  35. ^ a b Guckert, John (Spring 2012). "The Use of FFT and MDCT in MP3 Audio Compression" (PDF). University of Utah. Retrieved 14 July 2019.
  36. doi:10.1121/1.387544
    .
  37. ^ a b "Voice Coding for Communications". IEEE Journal on Selected Areas in Communications. 6 (2). February 1988.
  38. ^ a b c Genesis of the MP3 Audio Coding Standard in IEEE Transactions on Consumer Electronics, IEEE, Vol. 52, Nr. 3, pp. 1043–1049, August 2006
  39. ^ Brandenburg, Karlheinz; Seitzer, Dieter (3–6 November 1988). OCF: Coding High Quality Audio with Data Rates of 64 kbit/s. 85th Convention of Audio Engineering Society.
  40. doi:10.1109/49.608
    .
  41. ^ Y.F. Dehery, et al. (1991) A MUSICAM source codec for Digital Audio Broadcasting and storage Proceedings IEEE-ICASSP 91 pages 3605–3608 May 1991
  42. ^ "A DAB commentary from Alan Box, EZ communication and chairman NAB DAB task force" (PDF).
  43. ^ EBU SQAM CD Sound Quality Assessment Material recordings for subjective tests. 7 October 2008.
  44. ^
    Bloomberg BusinessWeek
    . Retrieved 24 July 2007.
  45. ISBN 978-0-14-310934-1. Brandenburg and Grill were joined by four other Fraunhofer researchers. Heinz Gerhauser oversaw the institute´s audio research group; Harald Popp
    was a hardware specialist; Ernst Eberlein was a signal processing expert; Jurgen Herre was another graduate student whose mathematical prowess rivaled Brandenburg´s own. In later years this group would refer to themselves as "the original six".
  46. ISBN 978-0-8223-5287-7
    .
  47. ^ "Suzanne Vega | Bio". The Official Community of Suzanne Vega. Retrieved 17 January 2022.
  48. ISBN 3-540-76357-0
    p. 144: "In the year 1988, the MASCAM method was developed at the Institut für Rundfunktechnik (IRT) in Munich in preparation for the digital audio broadcasting (DAB) system. From MASCAM, the MUSICAM (masking pattern universal subband integrated coding and multiplexing) method was developed in 1989 in cooperation with CCETT, Philips and Matsushita."
  49. ^ "Status report of ISO MPEG" (Press release). International Organization for Standardization. September 1990. Archived from the original on 14 February 2010.
  50. ^ "Aspec-Adaptive Spectral Entropy Coding of High Quality Music Signals". AES E-Library. 1991. Retrieved 24 August 2010.
  51. ^ "The MP3: A History Of Innovation And Betrayal". NPR. 23 March 2011. Retrieved 3 August 2023.
  52. ^ MPEG (25 March 1994). "Approved at 26th meeting (Paris)". Archived from the original on 26 July 2010. Retrieved 5 August 2010.
  53. ^ MPEG (11 November 1994). "Approved at 29th meeting". Archived from the original on 8 August 2010. Retrieved 5 August 2010.
  54. ^ ISO. "ISO/IEC TR 11172-5:1998 – Information technology – Coding of moving pictures and associated audio for digital storage media at up to about 1,5 Mbit/s – Part 5: Software simulation". Retrieved 5 August 2010.
  55. ^ "ISO/IEC TR 11172-5:1998 – Information technology – Coding of moving pictures and associated audio for digital storage media at up to about 1,5 Mbit/s – Part 5: Software simulation (Reference Software)" (ZIP). Retrieved 5 August 2010.
  56. ISBN 978-0-444-81580-4
    . This article refers to a Musicam (MPEG Audio Layer II) compressed digital audio workstation implemented on a microcomputer used not only as a professional editing station but also as a server on Ethernet for a compressed digital audio library, therefore anticipating the future MP3 on Internet
  57. ^ "MP3 Today's Technology". Lots of Informative Information about Music. 2005. Archived from the original on 4 July 2008. Retrieved 15 September 2016.
  58. ^ The heavenly jukebox on The Atlantic "To show industries how to use the codec, MPEG cobbled together a free sample program that converted music into MP3 files. The demonstration software created poor-quality sound, and Fraunhofer did not intend that it be used. The software's "source code"—its underlying instructions—was stored on an easily accessible computer at the University of Erlangen, from which it was downloaded by one SoloH, a hacker in the Netherlands (and, one assumes, a Star Wars fan). SoloH revamped the source code to produce software that converted compact-disc tracks into music files of acceptable quality." (2000)
  59. ^ Pop Idols and Pirates: Mechanisms of Consumption and the Global Circulation ... by Dr Charles Fairchild
  60. ^ Technologies of Piracy? - Exploring the Interplay Between Commercialism and Idealism in the Development of MP3 and DivX by HENDRIK STORSTEIN SPILKER, SVEIN HÖIER, page 2072
  61. Archive.org
    )
  62. ^ a b "Adopted at 22nd WG11 meeting" (Press release). International Organization for Standardization. 2 April 1993. Archived from the original on 6 August 2010. Retrieved 18 July 2010.
  63. ^ Brandenburg, Karlheinz; Bosi, Marina (February 1997). "Overview of MPEG Audio: Current and Future Standards for Low-Bit-Rate Audio Coding". Journal of the Audio Engineering Society. 45 (1/2): 4–21. Retrieved 30 June 2008.
  64. ^
    Fraunhofer IIS. September 2007. Archived from the original
    on 24 January 2008. "MPEG-2.5" is the name of a proprietary extension developed by Fraunhofer IIS. It enables MP3 to work satisfactorily at very low bitrates and introduces the additional sampling rates 8 kHz, 11.025 kHz and 12 kHz.
  65. ^ a b c d e f g h i Supurovic, Predrag (22 December 1999). "MPEG Audio Frame Header". Retrieved 29 May 2009.
  66. ^ a b c "ISO/IEC 13818-3:1994(E) – Information Technology — Generic Coding of Moving Pictures and Associated Audio: Audio" (ZIP). 11 November 1994. Retrieved 4 August 2010.
  67. ^ "About Internet Underground Music Archive".
  68. ^ Vainilavičius, Justinas (15 November 2023). "Winamp is back after revamp; nostalgia-inducing looks intact". cybernews. Retrieved 8 December 2023.
  69. ^ a b Schubert, Ruth (10 February 1999). "Tech-savvy Getting Music For A Song; Industry Frustrated That Internet Makes Free Music Simple". Seattle Post-Intelligencer. Retrieved 22 November 2008.[permanent dead link]
  70. S2CID 145796529
    .
  71. ^ a b c d Bouvigne, Gabriel (2003). "MP3 Tech — Limitations". Archived from the original on 7 January 2011.
  72. ^ "ISO/IEC 11172-3:1993/Cor 1:1996". International Organization for Standardization. 2006. Retrieved 27 August 2009.
  73. ^ Taylor, Mark (June 2000). "LAME Technical FAQ". Retrieved 9 December 2023.
  74. ^ Liberman, Serbio. DSP - The Technology Behind Multimedia.
  75. ^ Amorim, Roberto (3 August 2003). "Results of 128 kbit/s Extension Public Listening Test". Retrieved 17 March 2007.
  76. ^ Mares, Sebastian (December 2005). "Results of the public multiformat listening test @ 128 kbps". Retrieved 17 March 2007.
  77. ^ Dougherty, Dale (1 March 2009). "The Sizzling Sound of Music". O'Reilly Radar.
  78. ^ "Meet the Musical Clairvoyant Who Finds Ghosts In Your MP3s". NOISEY. 18 March 2015.
  79. ^ "The ghosts in the mp3". 15 March 2015. Archived from the original on 14 June 2017. Retrieved 25 April 2015.
  80. ^ "Lost and Found: U.Va. Grad Student Discovers Ghosts in the MP3". UVA Today. 23 February 2015.
  81. ^ The Ghost in the MP3
  82. ^ "Guide to command line options (in CVS)". Retrieved 4 August 2010.
  83. ^ "JVC RC-EX30 operation manual" (PDF) (in multiple languages). 2004. p. 14. Archived from the original (PDF) on 20 August 2020. Search – locating a desired position on thedisc (audio CD only) (2004 boombox)
  84. ^ "DV-RW250H Operation-Manual GB" (PDF). 2004. p. 33. • Fast forward and review playback does not work with a MP3/WMA/JPEG-CD.
  85. ^ "Sound Quality Comparison of Hi-Res Audio vs. CD vs. MP3". www.sony.com. Sony. Retrieved 11 August 2020.
  86. ISBN 978-0-471-73841-1
    .
  87. ^ Bouvigne, Gabriel (28 November 2006). "freeformat at 640 kbit/s and foobar2000, possibilities?". Retrieved 15 September 2016.
  88. ^ "lame(1): create mp3 audio files - Linux man page". linux.die.net. Retrieved 22 August 2020.
  89. ^ "Linux Manpages Online - man.cx manual pages". man.cx. Retrieved 22 August 2020.
  90. ^ a b "GPSYCHO – Variable Bit Rate". LAME MP3 Encoder. Retrieved 11 July 2009.
  91. ^ "TwoLAME: MPEG Audio Layer II VBR". Retrieved 11 July 2009.
  92. ^ ISO MPEG Audio Subgroup. "MPEG Audio FAQ Version 9: MPEG-1 and MPEG-2 BC". Retrieved 11 July 2009.
  93. ^ "LAME Y switch". Hydrogenaudio Knowledgebase. Retrieved 23 March 2015.
  94. ^ Rae, Casey. "Metadata and You". Future of Music Coalition. Archived from the original on 29 June 2017. Retrieved 12 December 2014.
  95. ^ Patel, Ketan; Smith, Brian C.; Rowe, Lawrence A. Performance of a Software MPEG Video Decoder (PDF). ACM Multimedia 1993 Conference.
  96. ^ "The MPEG-FAQ, Version 3.1". 14 May 1994. Archived from the original on 23 July 2009.
  97. ^ a b "A Big List of MP3 Patents (and supposed expiration dates)". tunequest. 26 February 2007.
  98. ^ Cogliati, Josh (20 July 2008). "Patent Status of MPEG-1, H.261 and MPEG-2". Kuro5hin. This work failed to consider patent divisions and continuations.
  99. ^ US Patent No. 5812672
  100. ^ "US Patent Expiration for MP3, MPEG-2, H.264". OSNews.com.
  101. ^ "Method and apparatus for encoding digital signals employing bit allocation using combinations of different threshold models to achieve desired bit rates".
  102. ^ "mp3licensing.com – Patents". mp3licensing.com.
  103. ^ "Full MP3 support coming soon to Fedora". 5 May 2017.
  104. ^ "Acoustic Data Compression – MP3 Base Patent". Foundation for a Free Information Infrastructure. 15 January 2005. Archived from the original on 15 July 2007. Retrieved 24 July 2007.
  105. Technicolor. Archived from the original
    on 4 May 2011.
  106. ^ Kistenfeger, Muzinée (July 2007). "The Fraunhofer Society (Fraunhofer-Gesellschaft, FhG)". British Consulate-General Munich. Archived from the original on 18 August 2002. Retrieved 24 July 2007.
  107. Chilling Effects Clearinghouse. 1 September 1998. Archived from the original
    on 19 August 2014. Retrieved 24 July 2007.
  108. ^ "SISVEL's MPEG Audio licensing programme".
  109. ^ "Audio MPEG and Sisvel: Thomson sued for patent infringement in Europe and the United States — MP3 players stopped by customs". ZDNet India. 6 October 2005. Archived from the original on 11 October 2007. Retrieved 24 July 2007.
  110. ^ "grants Motorola an MP3 and MPEG 2 audio patent license". SISVEL. 21 December 2005. Archived from the original on 21 January 2014. Retrieved 18 January 2014.
  111. ^ "US MPEG Audio patents" (PDF). Sisvel. Archived from the original (PDF) on 27 October 2016. Retrieved 7 April 2017.
  112. ^ "Licensing Programs - Legacy programs". www.sisvel.com. Archived from the original on 19 February 2023. Retrieved 15 September 2023.
  113. CNET News. Archived from the original
    on 4 November 2012. Retrieved 24 July 2007.
  114. ^ "Sisvel brings Patent Wild West into Germany". IPEG blog. 7 September 2006. Retrieved 24 July 2007.
  115. ^ "Apple, SanDisk Settle Texas MP3 Patent Spat". IP Law360. 26 January 2009. Retrieved 16 August 2010.
  116. Baker Botts LLP. Archived from the original
    on 10 December 2014. Retrieved 15 September 2016.
  117. ^ "Microsoft faces $1.5bn MP3 payout". BBC News. 22 February 2007. Retrieved 30 June 2008.
  118. ^ "Microsoft wins reversal of MP3 patent decision". CNET. 6 August 2007. Retrieved 17 August 2010.
  119. ^ "Court of Appeals for the Federal Circuit Decision" (PDF). 25 September 2008. Archived from the original (PDF) on 29 October 2008.
  120. ^ a b Brandenburg, Karlheinz (1999). "MP3 and AAC Explained". Archived from the original (PDF) on 19 October 2014.
  121. ^ "Via Licensing Announces Updated AAC Joint Patent License". Business Wire. 5 January 2009. Retrieved 18 June 2019.
  122. ^ "AAC Licensors". Via Corp. Archived from the original on 28 June 2019. Retrieved 6 July 2019.
  123. ^ Marriott, Michel (30 September 1999). "NEWS WATCH; New Player from Sony Will Give a Nod to MP3". The New York Times.
  124. ^ "Sony NW-E105 Network Walkman".
  125. arXiv:1602.04845
    . Its CBR produces packets with exactly the size the encoder requested, without a bit reservoir to imposes additional buffering delays, as found in codecs such as MP3 or AAC-LD. [...] [Tonal noise] is most noticeable in low-bitrate MP3s.

Further reading

External links

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