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MP3s
MPEG-1 Audio Layer 3, more commonly referred to as MP3s, is a patented digital audio encoding format using a form of lossy data compression. It is a common audio format for consumer audio storage, as well as a de facto standard of digital audio compression for the transfer and playback of music on digital audio players. MP3s is an audio-specific format that was designed by the Moving Picture Experts Group. The group was formed by several teams of engineers at Fraunhofer IIS in Erlangen, Germany, AT&T-Bell Labs in Murray Hill, NJ, USA, Thomson-Brandt, and CCETT as well as others. It was approved as an ISO/IEC standard in 1991.
The use in MP3s of a lossy compression algorithm is designed to greatly reduce the amount of data required to represent the audio recording and still sound like a faithful reproduction of the original uncompressed audio for most listeners. An MP3s file that is created using the mid-range bit rate setting of 128 kbit/s will result in a file that is typically about 1/10th the size of the CD file created from the original audio source. An MP3s file can also be constructed at higher or lower bit rates, with higher or lower resulting quality. The compression works by reducing accuracy of certain parts of sound that are deemed beyond the auditory resolution ability of most people. This method is commonly referred to as perceptual coding. It internally provides a representation of sound within a short term time/frequency analysis window, by using psychoacoustic models to discard or reduce precision of components less audible to human hearing, and recording the remaining information in an efficient manner. This is relatively similar to the principles used by JPEG, an image compression format.
When performing lossy audio encoding, such as creating an MP3s file, there is a trade-off between the amount of space used and the sound quality of the result. Typically, the creator is allowed to set a bit rate, which specifies how many kilobits the file may use per second of audio. Using a lower bit rate provides a relatively lower audio quality and produces a smaller file size. Likewise, using a higher bit rate outputs a higher quality audio, but also results in a larger file.
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 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 compressed with a relatively low bit rate provides a good example of compression artifacts.
Besides the bit rate of an encoded piece of audio, the quality of MP3s files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. As the MP3s standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two different MP3s encoders 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. However, in 1998, MP3s at 128 kbit/s was only providing quality equivalent to HE-AAC at 64 kbit/s and MP2 at 192 kbit/s.
The simplest type of MP3s file uses one bit rate for the entire file — this is known as Constant Bit Rate (CBR) encoding. Using a constant bit rate makes encoding simpler and faster. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate (VBR) files. The idea behind this is that, in any piece of audio, some parts will be much easier to compress, such as silence or music containing only a few instruments, 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 encoders, it is possible to specify a given quality, and the encoder will vary the bit rate accordingly. Users who know a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.
In a listening test, MP3s encoders at low bit rates performed significantly worse than those using more modern compression methods (such as AAC). In a 2004 public listening test at 32 kbit/s, the LAME MP3s encoder scored only 1.79/5 — behind all modern encoders — with Nero Digital HE AAC scoring 3.30/5.
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).
A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3s quality music has risen each year. Berger said the students seem to prefer the 'sizzle' sounds that MP3s bring to music. Others have reached the same conclusion, and some record producers have begun to mix music specifically to be heard on iPods and mobile phones. However, the study was criticized for being a short-term A/B test, which does not reflect the listeners preferences when they listen to music for prolonged periods.
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