MPEG-4 Part 14, formally, ISO/IEC 14496-14:2003, is a multimedia container format standard specified as a part of MPEG-4. It is most-commonly used to store digital audio and digital video streams, especially those defined by MPEG, but also can be used to store other data such as subtitles and still images. Like most modern container formats, MPEG-4 Part 14 allows streaming over the Internet. The official filename extension for MPEG-4 Part 14 files is .mp4, thus the container format is often referred to simply as MP4. Devices that play .mp4 files are referred to as MP4 players.
The existence of two different file extensions for naming audio-only MP4 files has been a source of confusion among users and multimedia playback software. Since MPEG-4 is a container format, MP4 files may contain any number of audio, video, and even subtitle streams, making it impossible to guess what type of streams are present in an MP4 file based on its filename extension alone. In response, Apple Computer started using and popularizing the .m4a file extension. Software capable of audio/video playback should recognize files with either .m4a or .mp4 file extensions, as would be expected, as there are no file format differences between the two. Most software capable of creating MPEG-4 audio will allow the user to choose the filename extension of the created MPEG-4 files.
Almost any kind of data can be embedded in MPEG-4 Part 14 files through private streams; the widely-supported codecs and additional data streams are:
Video: MPEG-4 Part 10 (also known as H.264 and MPEG-4 AVC), MPEG-4 Part 2, MPEG-2, and MPEG-1.
Audio: AAC (also known as MPEG-2 Part 7), MP3 (also known as MPEG-1 Audio Layer 3), MPEG-4 Part 3, MP2 (also known as MPEG-1 Audio Layer 2), MPEG-1 Part 3, CELP (speech), TwinVQ (very low bitrates), SAOL (MIDI).
Subtitles: MPEG-4 Timed Text (also known as 3GPP Timed Text).
Some private stream examples include Nero's use of DVD subtitles (Vobsub) in MP4 files. They are however not a part of the MPEG-4 file format standard and programs are not required to support them.
While the only official file extension defined by the standard is .mp4, various file extensions are commonly used to indicate intended content:
Audio-only MP4 files generally have a .m4a extension.
MP4 files with audio streams encrypted by FairPlay Digital Rights Management as sold through the iTunes Music Store use the .m4p extension.
Audio book and podcast files, which also contain metadata including chapter markers, images, and hyperlinks, can use the extension .m4a, but more commonly use the .m4b extension.
MP4 files with audio and video generally use the .mp4 and .m4v extensions, occasionally .mp4v.
3G mobile phones use 3GP, a simplified version of MPEG-4 Part 12 (a.k.a MPEG-4/JPEG2000 ISO Base Media file format, MPEG-4 Part 14 is a derivated standard from ISO Base file format too), with the .3gp and .3g2 extensions. These files also store non-MPEG-4 data (H.263, AMR, TX3G).
The common, but non-standard use of the extensions .m4a and .m4v is due to the popularity of Apple Computer's iPod and the iTunes Music Store.
Most video software supports MPEG-4 Part 14, such as:
Avidemuxiola
foobar2000
iTunes
KSP Sound Player
Media Player Classic
MPlayer
Nero Media Player
QuickTime Player
RealPlayer
VLC media player
Winamp
MPEG-1 Audio Layer 3, more commonly referred to as MP3, is a popular digital audio encoding and lossy compression format, 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. It was invented by a team of European engineers who worked in the framework of the EUREKA 147 DAB digital radio research program, and it became an ISO/IEC standard in 1991.
MP3 is an audio-specific compression format. It provides a representation of pulse-code modulation-encoded audio in much less space than straightforward methods, by using psychoacoustic models to discard components less audible to human hearing, and recording the remaining information in an efficient manner. Similar principles are used by JPEG, a lossy image compression format.
The MP3 format uses a hybrid transformation to transform a time domain signal into a frequency domain signal:
32-band polyphase quadrature filter
36 or 12 tap MDCT; size can be selected independently for sub-bands 0...1 and 2...31
Aliasing reduction postprocessing
MP3 audio can be compressed with several different bit rates, providing a range of tradeoffs between data size and sound quality.
The MPEG specifications support Advanced audio coding (AAC) from MPEG-4 as MP3's successor, although other new audio formats have also achieved similar usage levels. However, MP3's extreme popularity makes it secure in its dominant position for the near future, with support from a huge range of software and hardware, including portable MP3 players and even some DVD and CD players. The large MP3 collections that many individuals have amassed will also ensure its longevity, in the same way as with any physical medium.
MPEG-1 Audio Layer 2 encoding began as the Digital Audio Broadcast (DAB) project managed by Egon Meier-Engelen of the Deutsche Forschungs- und Versuchsanstalt für Luft- und Raumfahrt (later on called Deutsches Zentrum für Luft- und Raumfahrt, German Aerospace Center) in Germany. This project was financed by the European Union as a part of the EUREKA research program where it was commonly known as EU-147. EU-147 ran from 1987 to 1994.
In 1991, there were two proposals available: Musicam (known as Layer 2), and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (The Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated to the encoding of high quality compressed audio. The Musicam format, based on sub-band encoding, was a key to settle the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame). Its technology and ideas were fully incorporated into the definition of ISO MPEG Audio Layer I and Layer II and further on of the Layer III (MP3) format. Under the chairmanship of Professor Mussmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).
A working group consisting of Leon Van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) took ideas from Musicam and ASPEC, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.
All algorithms were approved in 1991, finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3, published in 1993. Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3, originally published in 1995.
Compression efficiency of encoders is typically defined by the bit rate because compression rate depends on the bit depth and sampling rate of the input signal. Nevertheless, there are often published compression rates that use the CD parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2x16 bit). Sometimes the Digital Audio Tape (DAT) SP parameters are used (48 kHz, 2x16 bit). Compression ratios with this reference are higher, which demonstrates the problem of the term compression ratio for lossy encoders.
Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess the MP3 compression algorithm. This song was chosen because of its softness and simplicity, making it easier to hear imperfections in the compression format during playbacks. Some have taken to jokingly refer to Suzanne Vega as "The mother of MP3". Some more serious and critical audio excerpts (glockenspiel, triangle, accordion, ...) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.
In October 1993, MP2 (MPEG-1 Audio Layer 2) files appeared on the Internet and were often played back using the Xing MPEG Audio Player, and later in a program for Unix by Tobias Bading called MAPlay, which was initially released on February 22, 1994 (MAPlay was also ported to Microsoft Windows).
Initially the only encoder available for MP2 production was the Xing Encoder, accompanied by the program CDDA2WAV, a CD ripper that transforms CD audio tracks to Waveform Audio Files.
The Internet Underground Music Archive (IUMA) is generally recognized as the start of the on-line music revolution. IUMA was the Internet's first high-fidelity music web site, hosting thousands of authorized MP2 recordings before MP3 or the web was popularized.
In the first half of 1995 through the late 1990s, MP3 files began flourishing on the Internet. MP3 popularity was mostly due to, and interchangeable with, the successes of companies and software packages like Nullsoft's Winamp (released in 1997), mpg123, and Napster (released in 1999). Those programs made it very easy for the average user to playback, create, share, and collect MP3s.
Controversies regarding peer-to-peer file sharing of MP3 files have spread widely in recent years — largely because high compression enables sharing of files that would otherwise be too large and cumbersome to easily share. Some major record companies reacted by filing a lawsuit against Napster, due to the vastly increased spread of MP3s through the Internet, to protect their copyrights (see also intellectual property).
Commercial online music distribution services (like the iTunes Music Store) usually prefer other/proprietary music file formats that support Digital Rights Management (DRM) to control and restrict the use of digital music. The use of formats that support DRM is in an attempt to prevent copyright infringement of copyright protected materials, but methods exist to defeat most protection schemes, although such methods are considered illegal in many countries.
The MPEG-1 standard does not include a precise specification for an MP3 encoder. The decoding algorithm and file format, as a contrast, are well defined. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information in the raw audio (or rather its MDCT representation in the frequency domain). During encoding 576 time domain samples are taken and are transformed to 576 frequency domain samples. If there is a transient 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient.
This is the domain of psychoacoustics: the study of subjective human perception of sounds.
As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. It must be kept in mind that an encoder that is proficient at encoding at higher bitrates (such as LAME, which is in widespread use for encoding at higher bitrates) is not necessarily as good at other, lower bitrates.
Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", meaning that the decompressed output they produce from a given MP3 file will be the same (within a specified degree of rounding tolerance) as the output specified mathematically in the ISO/IEC standard document. The MP3 file has a standard format which is a frame consisting of 384, 576, or 1152 samples (depends on MPEG version and layer) and all the frames have associated header information (32 bits) and side information (9, 17, or 32 bytes, depending on MPEG version and stereo/mono). The header and side information help the decoder to decode the associated Huffman encoded data correctly.
Therefore, for the most part, comparison of decoders is almost exclusively based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).
The bit rate is variable for MP3 files. The general rule is that more information is included from the original sound file when a higher bit rate is used, and thus the higher the quality during playback. In the early days of MP3 encoding, a fixed bit rate was used for the entire file.
Bit rates available in MPEG-1 Layer 3 are 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, and the available sampling frequencies are 32, 44.1 and 48 kHz. 44.1 kHz is almost always used (coincides with the sampling rate of compact discs), and 128 kbit/s has become the de facto "good enough" standard, although 192 kbit/s is becoming increasingly popular over peer-to-peer file sharing networks. MPEG-2 and [the non-official] MPEG-2.5 includes some additional bit rates: 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s while providing lower sampling frequencies (8, 11.025, 12, 16, 22.05 and 24 kHz).
Variable bit rates (VBR) are also possible. Audio in MP3 files is divided into frames (which have their own bit rate), so it is possible to change the bit rate dynamically as the file is encoded (although not originally implemented, VBR is in extensive use today). This technique makes it possible to use more bits for parts of the sound with higher dynamics (more sound movement) and fewer bits for parts with lower dynamics, further increasing quality and decreasing storage space. This method compares to a sound activated tape recorder that reduces tape consumption by not recording silence. Some encoders utilize this technique to a great extent.
Non-standard bitrates up to 640 kbit/s can be achieved with the LAME encoder and the --freeformat option, however few MP3 players can play those files.
Because MP3 is a lossy format, it is able to provide a number of different options for its "bit rate" — that is, the number of bits of encoded data that are used to represent each second of audio. Typically, rates chosen are between 128 and 320 kilobit per second. By contrast, uncompressed audio as stored on a compact disc has a bit rate of 1411.2 kbit/s (16 bits/sample × 44100 samples/second × 2 channels).
MP3 files encoded with a lower bit rate will generally play back at a lower quality. With too low a bit rate, "compression artifacts" (i.e., sounds that were not present in the original recording) may appear in the reproduction. A good demonstration of compression artifacts is provided by the sound of applause: it is hard to compress because of its randomness and sharp attacks. Therefore compression artifacts are audible as ringing or pre-echo.
As well as the bit rate of the encoded file, the quality of MP3 files depends on the quality of the encoder and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even when targeting similar bitrates. As an example, in a public collective test featuring two different MP3 encoders at about 128kbps, one scored 3.66 on a 1-5 scale, while the other scored only 2.22.
Quality is heavily dependent on the choice of encoder and encoding parameters. While quality around 128kbps was somewhere between annoying and acceptable with older encoders, modern MP3 encoders can provide very good quality at those bitrates , not statistically different from quality provided by AAC, the technical successor of MP3. However, in 1998, MP3 at 128kbps was only providing quality equivalent to AAC-LC at 96kbps and MP2 at 192kbps .
The transparency threshold of MP3 can be estimated to be at about 190kbps with good encoders. However, due to design limitations of the MP3 format, a few specific samples cannot be transparently encoded, even at those high bitrates.
At lower bitrates, the quality of MP3 quickly degrades, and is way behind AAC quality at 32kbps, as demonstrated by a collective listening test .
It is also important to note that perceived quality can be influenced by listening environment (ambient noise), listener attention, and listener training.
An MP3 file is made up of multiple MP3 frames which consist of the MP3 header and the MP3 data. This sequence of frames is called an Elementary stream. Frames are independent items: one can cut the frames from a file and an MP3 player would be able to play it. The MP3 data is the actual audio payload. 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 being used, hence MPEG-1 Audio Layer 3 or MP3. After this, the values will differ depending on the MP3 file. The range of values for each section of the header along with the specification of the header is defined by ISO/IEC 11172-3. Most MP3 files today contain ID3 metadata which precedes or follows the MP3 frames; this is also shown in the diagram.
There are several limitations inherent to the MP3 format that cannot be overcome by using a better encoder.
Newer audio compression formats such as Vorbis and AAC no longer have these limitations.
In technical terms, MP3 is limited in the following ways:
Bitrate is limited to a maximum of 320 kbit/s (although some encoders can create higher bitrates even though there is little-to-no support for these higher bitrate mp3s)
Time resolution can be too low for highly transient signals, causing some smearing of percussive sounds
Frequency resolution is limited by the small long block window size, decreasing coding efficiency
No scale factor band for frequencies above 15.5/15.8 kHz
Joint stereo is done on a frame-to-frame basis
Encoder/decoder overall delay is not defined, which means lack of official provision for gapless playback. However, some encoders such as LAME can attach additional metadata that will allow players that are aware of it to deliver gapless playback.
Nevertheless, a well-tuned MP3 encoder can perform competitively even with these restrictions
Many other lossy audio codecs exist, including:
MPEG-1/2 Audio Layer 2 (MP2), MP3's predecessor;
MPEG-4 AAC, MP3's successor, used by Apple's iTunes Music Store and iPod
Ogg Vorbis from the Xiph.org Foundation, a free software and patent free codec.
MPC, also known as Musepack (formerly MP+), a derivative of MP2;
mp3PRO from Thomson Multimedia combining MP3 with SBR;
AC-3, used in Dolby Digital and DVD;
ATRAC, used in Sony's Minidisc;
Windows Media Audio (WMA) from Microsoft.
QDesign, used in QuickTime at low bitrates;
AMR-WB+ Enhanced Adaptive Multi Rate WideBand codec, optimized for cellular and other limited bandwidth use;
RealAudio from RealNetworks, frequently in use for streaming on websites;
Speex, free software and patent free codec based on CELP specifically designed for speech and VoIP.
mp3PRO, MP3, AAC, and MP2 are all members of the same technological family and depend on roughly similar psychoacoustic models. The Fraunhofer Gesellschaft owns many of the basic patents underlying these codecs, with Dolby Labs, Sony, Thomson Consumer Electronics, and AT&T holding other key patents.
There are also some lossless audio compression methods used on the Internet. While they are not similar to MP3, they are good examples of other compression schemes available. These include:
FLAC stands for 'Free Lossless Audio Codec'
Monkey's Audio
SHN, also known as Shorten
TTA
Wavpack
Apple Lossless
Listening tests have attempted to find the best-quality lossy audio codecs at certain bitrates. At 128 kbit/s, Ogg Vorbis, AAC, MPC and WMA Pro tied for first place with LAME MP3 a little behind. At 64 kbit/s, AAC-HE and mp3pro performed marginally better than other codecs. At high bitrates (128 kbit/s+), most people do not hear significant differences. What is considered 'CD quality' is quite subjective.
Though proponents of newer codecs such as WMA and RealAudio have asserted that their respective algorithms can achieve CD quality at 64 kbit/s, listening tests have shown otherwise; however, the quality of these codecs at 64 kbit/s is definitely superior to MP3 at the same bitrate. The developers of the patent-free Ogg Vorbis codec claim that their algorithm surpasses MP3, RealAudio and WMA sound quality, and the listening tests mentioned above support that claim. Thomson claims that its mp3PRO codec achieves CD quality at 64 kbit/s, but listeners have reported that a 64 kbit/s mp3PRO file compares in quality to a 112 kbit/s MP3 file and does not come reasonably close to CD quality until about 80 kbit/s.
MP3, which was designed and tuned for use alongside MPEG-1/2 Video, generally performs poorly on monaural data at less than 48 kbit/s or in stereo at less than 80 kbit/s.