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A portable media player (PMP) or digital audio player (DAP) is a portable consumer electronics device capable of storing and playing digital media such as audio, images, and video files. The data is typically stored on a CD, DVD, flash memory, microdrive, or hard drive. Most portable media players are equipped with a 3.5 mm headphone jack, which users can plug headphones into, or connect to a boombox or hifi system. In contrast, analog portable audio players play music from non-digital media that use analog signal storage, such as cassette tapes or vinyl records.

Often mobile digital audio players are marketed and sold as “portable MP3 players”, even if they also support other file formats and media types. Increasing sales of smartphones and tablet computers have led to a decline in sales of portable media players, leading to some devices being phased out, though flagship devices like the Apple iPod and Sony Walkman are still in production. Portable DVD players are still manufactured by brands across the world.

1.1 Background Of The Study

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.  In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon.  Ernst Terhardt et al. created an algorithm describing auditory masking with high accuracy. 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.

The psychoacoustic masking codec was first proposed in 1979, apparently independently, by Manfred R. Schroeder, et al. from Bell Telephone Laboratories, Inc. in Murray Hill, New Jersey, and M. A. Krasner[18] both in the United States. Krasner was the first to publish and to produce hardware for speech (not usable as music bit compression), but the publication of his results as a relatively obscure Lincoln Laboratory Technical Report did not immediately influence the mainstream of psychoacoustic codec development. Manfred Schroeder was already a well-known and revered figure in the worldwide community of acoustical and electrical engineers, but his paper was not much noticed, since it described negative results due to the particular nature of speech and the linear predictive coding (LPC) gain present in speech.

Both Krasner and Schroeder built upon the work performed by Eberhard F. Zwicker in the areas of tuning and masking of critical frequency bands,[19][20] that in turn built on the fundamental research in the area from Bell Labs of Harvey Fletcher and his collaborators.[21] A wide variety of (mostly perceptual) audio compression algorithms were reported in IEEE‘s refereed Journal on Selected Areas in Communications.[22] That journal reported in February 1988 on a wide range of established, working audio bit compression technologies, some of them using auditory masking as part of their fundamental design, and several showing real-time hardware implementations.

The genesis of the MP3 technology is fully described in a paper from Professor Hans Musmann  who chaired the ISO MPEG Audio group for several years. The immediate predecessors of MP3 were “Optimum Coding in the Frequency Domain” (OCF), and Perceptual Transform Coding (PXFM).[25] These two codecs, along with block-switching contributions from Thomson-Brandt, were merged into a codec called ASPEC, which was submitted to MPEG, and which won the quality competition, but that was mistakenly rejected as too complex to implement. 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 56000 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. It was primarily designed for Digital Audio Broadcasting (digital radio) and digital TV, and its basic principles disclosed to the scientific community by CCETT (France) and IRT (Germany) in Atlanta during an IEEE-ICASSP conference. This codec incorporated into a broadcasting system using COFDM modulation was demonstrated on air and on the field  together with Radio Canada and CRC Canada during the NAB show (Las Vegas) in 1991. 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. Stoll (IRT Germany), later known as psychoacoustic model I) and a real time decoder using one Motorola 56001 DSP chip running an integer arithmetics software designed by Y.F. Dehery’s team (CCETT, France). The simplicity of the corresponding decoder together with the high audio quality of this codec using for the first time a 48 kHz sampling frequency, 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 a thorough use of a set of high quality audio assessment material [28] 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, ..) 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.[29] MP3 is directly descended from OCF and PXFM, representing the outcome of the collaboration of Brandenburg—working as a postdoc 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”[30]), 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 (in 1993 he joined the staff of the Fraunhofer Institute).[29] The song “Tom’s Diner” by Suzanne Vega was the first song used by Karlheinz Brandenburg to develop the MP3. Brandenburg adopted the song for testing purposes, listening to it again and again each time refining the scheme, making sure it did not adversely affect the subtlety of Vega’s voice (

1.2 Statement Of Problem

There are different problem faces by exiting media player we have to some will software some particular extension of music types where other won’t that will now rises for individual start installing different type of media player on their system or phone to be able to play different extension of video or audio they have, which will take a lot of spaces in there system or phone.

1.3 Aim Of The Study

The aim of this study is to design and implement of portable media player to organize individual music play list.

1.4 Objective Of The Study

The objective of this project is to:

  1. Focuses on program that have the main function of playing media
  2. Develop software that can play different extension of audio and video.
  3. Save time and reduces stress of selecting music one after other to media player.

1.5 Significance Of The Study

Projects provide a flexible framework for engaging students in exploring curricular topics and developing important 21st century skills, such as communication, teamwork, and technology skills. In addition, students are motivated by the fun and creative format and the opportunity to make new friends around the world. For every individual, an advance portable media player enables quick and easy management of music play list.

1.6 Scope Of The Study

The research will center on the design and implementation of a portable media player system for individual to use in managing their music play list.

1.7 Limitation Of The Study

Usually, every work has some limitations and this study is not exempted.

The two major limitations of this study are the high programming technique as well as financial constraints. The high programming technique constraint in Visual and Microsoft Access for database prevents the researcher to have an in depth study and analysis on the subject matter. While the issue of financial constraint limits the frequency of investigation to/from the institution toward gathering the necessary information relevant for the study.

1.8 Definition Of The Study

ALGORITHM:  An algorithm is a set of instruction that shows the step to follow in order to solve a problem.

DATA:  These are raw facts that are to be processed.

FLOWCHART: is the diagrammatic representation of sequence of events to be followed in solving problem.

FILES: Is a collect of related data record.

FIELD NAME:  This contains items about every record.

PROGRAM: Is a set of instructions of given for the computer to carryout a specific task.

INTEGER: Is referred to as a whole number.

STRING: Is referred to as a finite sequence of 3-ero or more character.

PERIPHERALS: is the physical component of the computer system.

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