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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. With regard to audio compression the aspect of the standard most apparent to end-users, and for which it is best known , MP3 uses lossy data-compression 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 lates, with MP3 serving as an enabling technology at a time when bandwidth and storage were still at a premium. With the advent of portable media players , a product category also including 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 as psychoacoustic modeling. The MP3 lossy audio-data compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking. In , the American physicist Alfred M. Mayer reported that a tone could be rendered inaudible by another tone of lower frequency.

Perceptual coding was first used for speech coding compression with linear predictive coding LPC , [22] which has origins in the work of Fumitada Itakura Nagoya University and Shuzo Saito Nippon Telegraph and Telephone in Atal and Manfred R. Schroeder at Bell Labs proposed an LPC speech codec , called adaptive predictive coding , that used a psychoacoustic coding-algorithm exploiting the masking properties of the human ear.

Hall was later reported in a paper. Krasner, [26] who published and produced hardware for speech not usable as music bit-compression , but the publication of his results in a relatively obscure Lincoln Laboratory Technical Report [27] did not immediately influence the mainstream of psychoacoustic codec-development.

The discrete cosine transform DCT , a type of transform coding for lossy compression , proposed by Nasir Ahmed in , was developed by Ahmed with T. Natarajan and K. Rao in ; they published their results in Princen, A. Johnson and A. Bradley in , [31] following earlier work by Princen and Bradley in Ernst Terhardt et al. In , Atal and Schroeder presented code-excited linear prediction CELP , an LPC-based perceptual speech-coding algorithm with auditory masking that achieved a significant data compression ratio for its time.

In June , 14 audio coding algorithms were submitted. Because of certain similarities between these coding proposals, they were clustered into four development groups. The first practical implementation of an audio perceptual coder OCF in hardware Krasner’s hardware was too cumbersome and slow for practical use , was an implementation of a psychoacoustic transform coder based on Motorola 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 filterbank, driven by a psychoacoustic model. The implementation of the audio part of this broadcasting system was based on a two-chips encoder one for the subband transform, one for the psychoacoustic model designed by the team of G. 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 [41] 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 the high-quality sound materials but especially for the encoding of critical percussive sound materials drums, triangle, As a doctoral student at Germany’s University of Erlangen-Nuremberg , Karlheinz Brandenburg began working on digital music compression in the early s, focusing on how people perceive music.

He completed his doctoral work in In , 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 , he joined the staff of Fraunhofer HHI. Brandenburg adopted the song for testing purposes, listening to it again and again each time he refined the scheme, making sure it did not adversely affect the subtlety of Vega’s voice.

This extension was developed at Fraunhofer IIS, the registered patent holders of MP3, by reducing the frame sync field in the MP3 header from 12 to 11 bits. 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. It is nonetheless ubiquitous and especially advantageous for low-bit-rate human speech applications.

Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published.

They may use the Compact Disc CD parameters as references Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.

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. Some other real time implementations of MPEG Audio encoders and decoders [58] were available for the purpose of digital broadcasting radio DAB , television DVB towards consumer receivers and set top boxes. A hacker named SoloH discovered the source code of the “dist10” MPEG reference implementation shortly after the release on the servers of the University of Erlangen.

He developed a higher-quality version and spread it on the internet. This code started the widespread CD ripping and digital music distribution as MP3 over the internet. In the second half of the s, MP3 files began to spread on the Internet , often via underground pirated song networks. After some experiments [64] using uncompressed audio files, this archive started to deliver on the native worldwide low-speed Internet some compressed MPEG Audio files using the MP2 Layer II format and later on used MP3 files when the standard was fully completed.

The popularity of MP3s began to rise rapidly with the advent of Nullsoft ‘s audio player Winamp , released in In November , the website mp3. The first large peer-to-peer filesharing network, Napster , was launched in The ease of creating and sharing MP3s resulted in widespread copyright infringement.

Major record companies argued that this free sharing of music reduced sales, and called it ” music piracy “.

They reacted by pursuing lawsuits against Napster , which was eventually shut down and later sold, and against individual users who engaged in file sharing. Unauthorized MP3 file sharing continues on next-generation peer-to-peer networks. An MP3 file is made up of MP3 frames, which consist of a header and a data block.

This sequence of frames is called an elementary stream. Due to the “bit reservoir”, frames are not independent items and cannot usually be extracted on arbitrary frame boundaries. The MP3 Data blocks contain the compressed audio information in terms of frequencies and amplitudes.

The diagram shows that the MP3 Header consists of a sync word , which is used to identify the beginning of a valid frame. After this, the values will differ, depending on the MP3 file. The data stream can contain an optional checksum. Joint stereo is done only on a frame-to-frame basis.

The MP3 encoding algorithm is generally split into four parts. Part 1 divides the audio signal into smaller pieces, called frames, and a modified discrete cosine transform MDCT filter is then performed on the output. Part 2 passes the sample into a point fast Fourier transform FFT , then the psychoacoustic model is applied and another MDCT filter is performed on the output. Part 3 quantifies and encodes each sample, known as noise allocation, which adjusts itself in order to meet the bit rate and sound masking requirements.

Part 4 formats the bitstream , called an audio frame, which is made up of 4 parts, the header , error check , audio data , and ancillary data. The MPEG-1 standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard.

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 LAME were not necessarily as good at lower bit rates. Later an ABR mode was added.

Work progressed on true variable bit rate using a quality goal between 0 and Eventually numbers such as -V 9. During encoding, time-domain samples are taken and are transformed to frequency-domain samples. This is done to limit 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. Decoding, on the other hand, is carefully defined in the standard. Therefore, comparison of decoders is usually based on how computationally efficient they are i.

However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback. When performing lossy audio encoding, such as creating an MP3 data stream, there is a trade-off between the amount of data generated and the sound quality of the results.

The person generating an MP3 selects 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. 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 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.