when we copy a music to the cd,we will be having two options.
we can do it in mp3 form. what does it mean.
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MP3 is a format for compressing audio files including music. It allows you to fit a lot of audio (which usually takes up a lot of space) into a much smaller space, by removing the bits of the sound that you're least likely to miss.
MP3 files can be played on almost all computers, portable media devices, and even many DVD players, so they're a good choice of compression format. Other compressed audio codecs such as WMA can include copyright information that makes it much more difficult to use your compressed music files on different devices, and are also less widely supported than MP3.
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MPEG-1 Audio Layer3, more commonly refferred to as MP3,is a popular digital audio encoding, lossu compession formal, and algorithm 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 of Philips, CCETT,IRT and Fraunhofer Society, who worked in the frame work of the EUREKA147DAB digital radio research program, and it became an ISO/IEC standard in 1991.
motion pictures experts group-3 (MPEG-3)Audio Layer 3, more commonly referred to as MP3, is a popular digital audio encoding, lossy compression format, and algorithm, 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 listeners.
MP3 is an audio-specific compression format. It provides a representation of pulse-code modulation鈥揺ncoded 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
When creating an MP3 file, there is a trade-off between the amount of space used and the sound quality of the result. Typically, the user is allowed to set a bit rate which specifies how many kilobits the file may use per second of audio. The lower the bit rate used, the lower will be the audio quality. Likewise, the higher the bitrate used, the higher quality the resulting MP3 will be.
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 be audible 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 can be heard 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 bit rates. As an example, in a public listening test featuring two different MP3 encoders at about 128kbps[2], one scored 3.66 on a 1鈥? 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 [3] (01/2006), 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[4].
The transparency threshold of MP3 can be estimated to be at about 128k with good encoders on typical music as evidenced by its strong performance in the above test, however some particularly difficult material can require 192k or higher. As with all lossy formats, some samples can not be encoded to be transparent for all users.
The simplest type of MP3 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 bitrate accordingly. Users who know a particular "quality setting" which 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 settings.
In listening tests, MP3 encoders have performed significantly worse than those using more modern compression methods (such as AAC) at low bit rates. In a 2004 public listening test at 32 kbit/s[5], the LAME MP3 encoder scored only 1.79/5 - behind all modern encoders - with Nero Digital HE AAC scoring 3.30/5.
It is also important to note that perceived quality can be influenced by listening environment (ambient noise), listener attention, and listener training.
[edit] Bit rate
Several bit rates are specified in the MPEG-1 Layer 3 standard: 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. A sample rate of 44.1 kHz is almost always used since this is also used for CD audio, the main source used for creating MP3 files. A greater variety of bitrates are used on the internet. 128 kbit/s is the most common since it typically offers very good audio quality in a relatively small space. 192 kbit/s is often used by those who notice artifacts at lower bitrates. By contrast, uncompressed audio as stored on a compact disc has a bit rate of 1411.2 kb/s (16 bits/sample 脳 44100 samples/second 脳 2 channels).
Some additional bit rates and sample rates were made available in the MPEG-2 and the (unofficial) MPEG-2.5 standards: bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kb/s and sample rates of 8, 11.025, 12, 16, 22.05 and 24 kHz.
Non-standard bit rates up to 640 kb/s can be achieved with the LAME encoder and the --freeformat option, but few MP3 players can play those files. Gabriel Bouvigne, a principal developer of the LAME project, says that the freeformat option is compliant with the standard but, according to the standard, decoders are only required to be able to decode streams up to 320 kbit/s.[6]
[edit] File structure
MP3 File StructureAn 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. 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; this is also shown in the diagram.
[edit] Design limitations
There are several limitations inherent to the MP3 format that can not be overcome by any MP3 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:
Bit rate is limited to a maximum of 320 kb/s (while some encoders can create higher bit rates, there is little-to-no support for these higher bit rate 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 seamless playback.
Nevertheless, a well-tuned MP3 encoder can perform competitively even with these restrictions.
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