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MP3 Converter:

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Details About MP3 and MP3 Converter:

tuneCopy MP3 Converter excels as the most versatile and powerful MP3 Converter. It can convert any music files into MP3 format.

The MP3 format uses, at its heart, a hybrid transform 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 independent for sub-band 0...1 and 2...31
* aliasing reduction postprocessing

MP3 Surround, a version of the format supporting 5.1 channels for surround sound, was introduced in December 2004. MP3 Surround is backward compatible with standard stereo MP3, and file sizes are similar.

In terms of the MPEG specifications, AAC (Advanced audio coding) from MPEG-4 is to be the successor of the MP3 format, although there has been a significant movement to create and popularize other audio formats. Nevertheless, any succession is not likely to happen for a significant amount of time due to MP3's overwhelming popularity (MP3 enjoys extremely wide popularity and support, not just by end-users and software but by hardware such as DVD and CD players).

History::

Development

MPEG-1 Audio Layer 2 encoding began as the Digital Audio Broadcast (DAB) project initiated by the Fraunhofer Society 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 II), and ASPEC (Adaptive Spectral Perceptual Entropy Coding) with similarities to MP3. Musicam was chosen due to its simplicity and error robustness.

A working group around Karlheinz Brandenburg and Jürgen Herre 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.

Both algorithms were 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 which 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 for this reference is 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 as his model for 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.

MP3 went public

On July 7, 1994 the Fraunhofer Society released the first software MP3 encoder called l3enc. The filename extension .mp3 was chosen by the Fraunhofer team on July 14, 1995 (previously, the files had been named .bit). With the first realtime software MP3 player Winplay3 (released September 9th, 1995) many people were able to encode and playback MP3 files on their PCs. Because of the relatively small hard drives back in that time (~500 MB) the technology was essential to store music for listening pleasure on a computer.

MP2 and MP3 and the Internet

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 22nd, 1994 (MAPlay was also ported to the Microsoft Windows OS).

Initially the only encoder available for MP2 production was the Xing Encoder, accompanied by the program CDDA2WAV, a CD ripper that transformed CD audio tracks to computer data 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 were popularized. IUMA was started by Rob Lord (who later headed pioneering Nullsoft) and Jeff Patterson, both from the University of California, Santa Cruz, in 1993. Other founding members include Jon Luini, Brandee Selck, and Ahin Savara.

In the first half of 1995, 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, mpg123, and (now Roxio-owned) Napster. 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 has flourished in recent years — largely because high compression enables sharing of files that would otherwise be too large and cumbersome to share. Due to the vastly increased spread of MP3s through the internet some major record labels reacted by filing a lawsuit against Napster 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 DRM to control and restrict the use of digital music. This preference is most likely chosen in an attempt to prevent piracy of copyrighted materials, but most users with at least an intermediate understanding of computers will know that it's just a matter of time before someone else makes it easy to convert such proprietary file formats.

Quality of MP3 audio

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 256 kibibits per second. By contrast, uncompressed compact disc audio has a bit rate of (1378 kibits/s?) (1411 kibits/s?).

Looking at bit rate of CD audio from the sector perspective: Bit rate of CD-DA (audio CD) = 2352 bytes/sector x 75 sectors/s = 176,400 bytes/s = 172.27 KB/s = 0.17 MB/s = about 10 MB per minute.*

Looking at bit rate of CD audio from the sampling rate perspective: CD audio is sampled using PCM to 16 bits per channel, with two channels, at 44.1 kHz. Therefore, bit rate = (44100 samples/channel)/s x 16 bits/sample x 2 channels = 1411200 bits/s = 176400 bytes/s = 172.27 KB/s = 0.17 MB/s = about 10 MB per minute.*

* Actual bit rate is higher, because of EFM, CIRC, L2 ECC, and so on.

(For purposes of comparison, bit rate of data CD = 2048 bytes/sector x 75 sectors/s = exactly 150 KB/s = about 8.8 MB per minute.)

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 it is random, therefore the failings of the encoder are more obvious, and are audible as ringing.

As well as the bit rate of the encoded file, the quality of MP3 files depend on the quality of the encoder and the difficulty of the signal being encoded. For average signals with good encoders, many listeners accept the MP3 bit rate of 128 kibit/s as near enough to compact disc quality for them, providing a compression ratio of approximately 11:1. However, listening tests show that with a bit of practice many listeners can reliably distinguish 128 kibit/s MP3s from CD originals; in many cases reaching the point where they consider the MP3 audio to be of unacceptably low quality. Yet other listeners, and the same listeners in other environments (such as in a noisy moving vehicle or at a party) will consider the quality acceptable.

Fraunhofer Gesellschaft (FhG) publish on their official webpage the following compression ratios and data rates for MPEG-1 Layer 1, 2 and 3, intended for comparison:

* Layer 1: 384 kbit/s, compression 4:1
* Layer 2: 192...256 kbit/s, compression 6:1...8:1
* Layer 3: 112...128 kbit/s, compression 10:1...12:1

The differences between the layers are caused by the different psychoacoustic models used by them; the Layer 1 algorithm is typically substantially simpler, therefore a higher bit rate is needed for transparent encoding. However, as different encoders use different models, it is difficult to draw absolute comparisons of this kind.

Many people consider these quoted rates as being heavily skewed in favour of Layer 2 and Layer 3 recordings. They would contend that more realistic rates would be as follows:

* Layer 1: excellent at 384 kbit/s
* Layer 2: excellent at 256...384 kbit/s, very good at 224...256 Kbit/s, good at 192...224 Kbit/s
* Layer 3: excellent at 224...320 Kbit/s, very good at 192...224 Kbit/s, good at 128...192 Kbit/s

When comparing compression schemes, it is important to use encoders that are of equivalent quality. Tests may be biased against older formats in favour of new ones by using older encoders based on out-of-date technologies, or even buggy encoders for the old format. Due to the fact that their lossy encoding loses information, MP3 algorithms work hard to ensure that the parts lost cannot be detected by human listeners by modeling the general characteristics of human hearing (e.g., due to noise masking). Different encoders may achieve this with varying degrees of success.

A few possible encoders:

* LAME first created by Mike Cheng in early 1998. It is (in contrast to others) a fully LGPL'd MP3 encoder, with excellent speed and quality, rivaling even MP3's technological successors.
* Fraunhofer Gesellschaft: Some encoders are good, some have bugs.

Many early encoders are no longer widely used:

* ISO dist10 reference code
* Xing
* BladeEnc
* ACM Producer Pro.

Good encoders produce acceptable quality at 128 to 160 Kibit/s and near-transparency at 160 to 192 Kibit/s, while low quality encoders may never reach transparency, not even at 320 Kbit/s. It is therefore misleading to speak of 128 Kibit/s or 192 Kibit/s quality, except in the context of a particular encoder or of the best available encoders. A 128 Kibit/s MP3 produced by a good encoder might sound better than a 192 Kibit/s MP3 file produced by a bad encoder.

It is important to note that quality of an audio signal is subjective. A given bit rate suffices for some listeners but not for others. Individual acoustic perception may vary, so it is not evident that a certain psychoacoustic model can give satisfactory results for everyone. Merely changing the conditions of listening, such as the audio playing system or environment, can expose unwanted distortions caused by lossy compression. The numbers given above are rough guidelines that work for many people, but in the field of lossy audio compression the only true measure of the quality of a compression process is to listen to the results.

If your aim is to archive sound files with no loss of quality (or work on the sound files in a studio), then of interest is Lossless compression algorithms that are currently capable of compressing 16-bit PCM audio by 38 to 80% (partly depending upon the characteristics of the audio itself) while leaving the audio identical to the original, such as Free Lossless Audio Codec (FLAC) or Apple Lossless (among others). Lossless formats are strongly preferred for material which will be edited, mixed, or otherwise processed because the perceptual assumptions made by lossy coders may no long hold true after processing, and because the losses produced by multiple stages of coding may compound each other, becoming more evident when the signal is reencoded after processing. Lossless formats produce the best possible result, at the expense of a lower compression ratio.

Some simple editing operations, such as cutting sections of audio, may be performed directly on the encoded MP3 data without necessitating reencoding. For these operations, the concerns mentioned above are not necessarily relevant, as long as appropriate software (such as mp3DirectCut and MP3Gain) is used to prevent extra decoding-encoding steps.

Bit rate

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 play back. 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 kibit/s, and the available sample 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 Kibit/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 Kibit/s

Variable bit rates (VBR) are also possible. Audio in MP3 files are 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 which reduces tape consumption by not recording silence. Some encoders utilize this technique to a great extent.

Design limitations of MP3

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, AAC, Musepack and WMA no longer have these limitations.

In technical terms, MP3 is limited in the following ways:

* Bitrate is limited to a maximum of 320 kibit/s
* Time resolution can be too low for highly transient signals
* Encoder/decoder overall delay is not defined
* No scalefactor band for frequencies above 15.5/15.8 kHz
* Joint stereo is done on a frame-to-frame basis
* Lack of gapless playback; tracks intendended to flow into another (such as in live performances) do not seamlessly flow: a short gap of silence is heard between the two tracks.

Nevertheless, a well tuned MP3 encoder can perform competitively even with these restrictions.

Encoding of MP3 audio

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. Implementors 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). This is the domain of psychoacoustics, which aims at understanding how human acoustical perception works (both in our ears and in our brain).

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 of MP3 audio

Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", meaning that the uncompressed 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 standard document. 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).

ID3 and other tags

{{main|ID3]] and [[APEv2 tag}}

A "tag" is data stored in an MP3 (as well as other formats) which contains metadata such as the title, artist, album, track number or other information about the MP3 file to be added to the file itself. The most widespread standard tag formats are currently the ID3 ID3v1 and ID3v2 tags, and the more recent APEv2 tag.

APEv2 was originally developed for the MPC file format (see the APEv2 specification). APEv2 can coexist with ID3 tags in the same file, but it can also be used by itself.

Volume normalization

As compact discs and other various sources are recorded and mastered at different volumes, it is useful to store volume information about a file in the tag so that at playback time, the volume can be dynamically adjusted.

A few standards for encoding the gain of an MP3 file have been proposed. The idea is to normalize the volume (not the volume peaks) of audio files, so that the volume does not change between consecutive tracks.

The most popular and widely-used solution for storing replay gain is known simply as "Replay Gain". Typically, the average volume and clipping information about an audio track is stored in the metadata tag.

Alternative technologies

Many other lossy audio codecs exist, including:

* MPEG-1/2 Audio Layer 2 (MP2), MP3's predecessor;
* 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;
* MPEG-4 AAC, used by Apple's iTunes Music Store and iPod
* 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 non-lossy (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

Listening tests have attempted to find the best-quality lossy audio codecs at certain bitrates. The tests have suggested that for some audio samples, newer audio codecs including Ogg Vorbis, mp3PRO, AC-3, Windows Media Audio, MPC and RealAudio perform better than MP3. Generally, these codecs achieve the equivalent of MP3 128kbit/s at around 80kbit/s. At 128kbit/s, Ogg Vorbis and MPC performed marginally better than other codecs. At 64kbit/s, AAC and mp3pro performed marginally better than other codecs. At high bitrates (128kbit/s+), most people do not hear significant differences. What is considered 'CD quality' is quite subjective; for some 128kbit/s MP3 is sufficient, while for others 192kbit/s MP3 is necessary.

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 bandwidth. 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.

Licensing and patent issues

Thomson Consumer Electronics controls licensing of the MPEG-1/2 Layer 3 patents in countries that recognize software patents, including the United States, Japan, and most EU countries. Thomson has been actively enforcing these patents.

In September 1998, the Fraunhofer Institute sent a letter to several developers of MP3 software stating that a license was required to "distribute and/or sell decoders and/or encoders". The letter claimed that unlicensed products "infringe the patent rights of Fraunhofer and THOMSON. To make, sell and/or distribute products using the [MPEG Layer-3] standard and thus our patents, you need to obtain a license under these patents from us."

These patent issues significantly slowed the development of unlicensed MP3 software and led to increased focus on creating and popularising alternatives such as WMA and Ogg Vorbis. Microsoft, the makers of the Windows operating system, chose to move away from MP3 to their own proprietary Windows Media formats to avoid the licensing issues associated with the patents. Until the key patents expire, open source / free software encoders and players appear to be illegal for commercial use in countries that recognize software patents.

For information about licensing fees see here and here.

In spite of the patent restrictions, the perpetuation of the MP3 format continues; the reasons for this appear to be the network effects caused by:

* familiarity with the format, not knowing alternatives exist,
* the large quantity of music now available in the MP3 format,
* the wide variety of existing software and hardware that takes advantage of the file format.
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