Home : Audio Guide

Help and Guides

References:

• Chris Myden
  • Guide To Creating Amazing MP3s
    EAC is the best ripper. LAME is the best MP3 encoder. Want to install them both quickly, with all the best options? This straight-forward, no-nonsense, step-by-step guide is just for you.
  • Guide To Creating Amazing MPCs
    MPC, formerly known as mpegplus, is an excellent audio compression format. In fact, many say it is the only truly transparent form of lossy compression.
  • Guide To Creating Amazing OGGs
    OGG appears to be the front-runner for the heir to the MP3 crown. It is a completely open and free compression format and already has support from the hardware industry.
  • Using EAC Answers to all your questions about the greatest ripper available.
• DivX Digest
  • AVI Audio Normalize/Increase Volume Guide
    Find out how to increase and/or normlize the volume of AVI files.
  • WMA (DivX Audio) codec with any bitrate
    A guide that tells you how to use the WMA (DivX Audio) codec using any bitrate (not just the default 44.1 kHz 64kbps) - written by "Abducted from the Netherlands".
• Doom9 Networks
• LI Utilities
  • Reducing gaps and jumps when playing MP3 music
• Nicky
  • How to convert MP2 to PCM Wave
    Are some of your DVDs come with MPEG-2 audio? Can't convert it directly with FlasK or AC3Dec to WAV? Then the answer to your problem is this guide.
• VCDHelp.us
  • How to rip audio from DivX/AVI to WAV file

What is MPEG ?

MPEG which stands for Moving Picture Experts Group, is the name of familly of standards (joint of the International Organization for Standardization [ISO]) used for coding audio and video data in a digital compressed format including data transmission across digital networks.
MPEG Video files have the .mpg or .dat extension and MPEG Audio files generally have the extension .mp1, .mp2, .mp3. MPEG is cross-platform compatible and can be played on all popular computer systems.

The major advantage of MPEG compared to other video and audio coding formats is that MPEG files are much smaller for the same quality (eg, mp3 size is about 1:10 compared to Audio CD). This is because MPEG uses very sophisticated compression techniques.

-- www.xingtech.com
-- www.mpeg.org

What is CDDB ?

CDDB is the "Compact Disc Database". It is simply a huge collection of files, each of which contains a disc name, song titles and other information for a particular CD. CDDB now supports hundreds of different third party software CD players. Access to the CDDB service is offered free of charge to end-users.

-- www.cddb.com

What is MP3 ?

MPEG-1 Audio Layer 3, more commonly referred to as MP3, is a popular digital audio encoding and lossy compression format, 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 who worked in the framework of the EUREKA 147 DAB digital radio research program, and it became an ISO/IEC standard in 1991. The other advantages of MP3 is that it can be broken up into pieces, and each piece is still playable. The feature that makes this possible (headerless file format) also means that MP3 files can be made to stream across the net real-time (assuming the playback bitrate and speed of the Internet connection are compatible).
MP3 is an audio-specific compression format. It provides a representation of pulse-code modulation-encoded 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.
The MP3 format uses a hybrid transformation 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 independently for sub-bands 0...1 and 2...31
* Aliasing reduction postprocessing

MP3 audio can be compressed with different bit rates, providing a range of tradeoffs between data size and sound quality.
The MPEG specifications support Advanced audio coding (AAC) from MPEG-4 as MP3's successor, although other new audio formats have also achieved similar usage levels. However, MP3's extreme popularity makes it secure in its dominant position for the near future, with support from a huge range of software and hardware, including portable MP3 players and even some DVD and CD players. The large MP3 collections that many individuals have amassed will also ensure its longevity, in the same way as with any physical medium.

-- http://en.wikipedia.org/wiki/MP3

What is bitrate in MP3 ?

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 playback. 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 kbit/s, and the available sampling 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 kbit/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 kbit/s while providing lower sampling frequencies (8, 11.025, 12, 16, 22.05 and 24 kHz).
Variable bit rates (VBR) are also possible. Audio in MP3 files is 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 that reduces tape consumption by not recording silence. Some encoders utilize this technique to a great extent.

-- http://en.wikipedia.org/wiki/MP3

What does MPEG Audio Layer III refer to ?

Both MPEG-1 Audio and MPEG-2 Audio include a family of three closely related methods of compressing audio. These methods are known as Layer I, Layer II and Layer III. The MPEG Audio Layer III in MP3 refers to the use of the most advanced of these compression methods, Layer III.

Each of the three layers has its advantages in different situations, and that's why the MPEG Audio standars define three methods instead one. Of the three methods, Layer I compresses audio the least, but it requires the simplest software for creating and listening to the audio. Layer II is able to compress the audio to a greater degree, but it also requires more complex software, particulary in the creation of the compressed file. Because the software is more complex, this also means that the compression process is slower and requires a more powerfull computer. Layer III is the best of the three methods as far as compression is concerned, and as a result requires the most complex software.

Layer III has become the most popular of the three for digital music because of its greater compression, which made it much more practical for downloading and storing. This led to most software for creating digital audio being focused on Layer III, though most MP3 playing software can also play Layer I and Layer II files.
The Layer III compression scheme was developed in large part by researches at the Fraunhofer Institute for Integrated Circuits in Germany and was selected by the MPEG organization to be incorporated into its standards.

-- www.iis.fhg.de/amm/technif/layer3/

Is MPEG-3 the same as MP3 ?

Although it sounds similar, MPEG-3 isn't the same as MP3. If it had been completed, MPEG-3 would have been the next standard in the series following MPEG-1 and MPEG-2. MPEG-3 was intended to be use to define a transmission format for high definition TV (HDTV), but it was discovered that MPEG-2 standard could be used for that purpose, making a new MPEG-3 standard unnecessary.

-- www.muskalipman.com [MP3 FYI]

What is MPEG-2 AAC ?

MPEG-2 AAC is the consequent continuation of the truly successful coding method ISO/MPEG Audio Layer-3 developed in Erlangen. The extensive co-operation with international partners and the insight derived from Layer-3 paved the way for this coding method which is unprecedented at present stage. The appropriate incorporation of high coding gain and great flexibility opens up a wide field of applications. With sampling frequencies between 8 kHz and 96 kHz and any number of channels between 1 and 48, the method is well prepared for future developments in the audio sector. Compared to well-known coding methods such as MPEG-2 Layer-2, it is possible to achieve half the bit rate with no loss of subjective quality

Is MP3 legal ?

Yes. It can be used legally or illegally. Using an MP3 is legal if the song's copyright holder has granted permission to download and play the song. Also, it is legal to make copies from CD for personal use. But, it is illegal to encode MP3s from CD and trade them without permission of the copyright holder.

How to make MP3 files ?

1. Download any Audio Ripper software and any Audio Encoders software. Or you can download Audio Multi which contains both of them.
2. If your CD-ROM is compatible then you can use CD-Ripper to grab the audio files on the CD (.cda) to wave (.wav) format. Grabbing speed depends on your CD-ROM capability and your processor power. jump to number 4
3. If your CD-ROM is uncompatible or unsupported then you should record it manually using your audio software installed (bundled with the Audio card that plugged in your computer). By using the WAV recorder. The speed is fully depend on the length of a song (1:1).
4. Use an Audio Encoder software to convert the wave (.wav) files to .mp3 format (160 bit/s is now the standard bitrate on the net, higher is better).
5. Done. Now you can listen to the result by using Audio Player software.

Audio Dictionaries

• a2b music

A proprietary, secure music distribution system developed by AT&T. It uses AAC for compression.
• AAC

Advanced Audio Coding (AAC) is a standardised, lossy digital audio compression scheme. It was developed with the cooperation and contributions of companies mainly including Dolby, Fraunhofer (FhG), AT&T, Sony and Nokia, and was officially declared an international standard by the Moving Pictures Experts Group in April of 1997. It was written into specification as Part 7 of the MPEG-2 standard, and again into Part 3 of the MPEG-4 standard. As such, AAC can be referred to as MPEG-2 Part 7 and MPEG-4 Part 3 depending on its implementation, but is most often referred to as MPEG-4 AAC, or AAC for short.
AAC was designed as an improved-performance codec relative to MP3 (which was specified in MPEG-1 and MPEG-2) by the ISO/IEC in 11172-3 and 13818-3.
AAC was promoted as the successor to MP3 for audio coding at medium to high bitrates. Its popularity is currently maintained by it being the default iTunes codec, the media player which powers iPod, the most popular digital audio player on the market. Furthermore, the iTunes Store, whose sales account for 85% of the market for legal online downloads, sells AAC-encoded songs (encapsulated with FairPlay Digital Rights Management).
• AC3

AC-3 is a high-quality, low-complexity multichannel audio compression system proposed by the Dolby Laboratories. AC-3 is approved by the ATSC and free specifications are available. AC-3 is also called Dolby Digital 5.1, the six channels are Left, Right, Center, Surround Left, Surround Right, Low Frequency Effects (<120Hz). The total encoded bit stream fits in 384Kbps. Also known as Dolby Digital, it’s a digital audio format from Dolby Labs™. It can support up to 5.1 channels of audio.
• AIFF-C

Audio Interchange File Format (AIFF) is an audio file format standard used for storing sound data on personal computers. The format was co-developed by Apple Computer based on Electronic Arts Interchange File Format (IFF) and is most commonly used on Apple Macintosh computer systems. AIFF is also used by Silicon Graphics Incorporated.
The audio data in an AIFF file are uncompressed big-endian pulse-code modulation (PCM) so the files tend to be much larger than files that use lossless compression (such as FLAC) or lossy compression formats such as Vorbis and MP3. Uncompressed AIFF files at compact-disc settings (44.1K samples/sec, 16 bits, 2 channels) thus have a bitrate of 1411.2 kbit/s. The AIFF-Compressed (AIFF-C or AIFC) format supports compression ratios as high as 6:1.
• Analog Audio

A continuously varying representation of sound waves. In most analog audio systems, a continuously varying voltage signal represents the sound wave.
• AHRA

Audio Home Recording Act of 1992 allows comsumers to create copies of recordings for personal, non-commercial use.
• ATRAC3

Sony's Adaptive Transform Acoustic Coding 3 format is an improved version of the ATRAC encoding shceme used by MiniDisc Players. It archieves nearly the same level of compression and sound quality as MP3 at the same bitrate. Currently, the ATRACT3 format is primarly used by Sony's portable players. Sony said that ATRAC3 will be SDMI compliant.
• ATSC

The Advanced Television Systems Committee (ATSC) is an international organization of 200 members that is establishing voluntary technical standards for advanced television systems. ATSC Digital TV Standards include digital high definition television (HDTV), standard definition television (SDTV), data broadcasting, multichannel surround-sound audio, and Satellite direct-to-home broadcasting.
• AU

The Au file format is a simple audio file format that consists of a header of 6 32-bit words and then the data (high-order byte comes first). The format was introduced by Sun Microsystems.
• Bitrate

The number of bits per given time interval, used as a measure of information flow. Digital audio bitrate is often given in units of kbps (kilo bits per second).
• Buffer

A temporary storage of space used as a reserve to insure a smooth flow of data.
• CBR

Constant BitRate Encoding, the more commonly used method that allocates tht same number of bits per second of audio throughout the entire track. The advantage of this method is that the size of the finished MP3 can be calculated exactly before encoding it just by knowing the length of the track and the bitrate it will be encoded at. The main disadvantage of CBR is that it requires the encoder to use the same amount of information to represent both complex and very simple sections of audio. In a simple section of audio, such as silence, the encoder must waste bits, although it might not have enough bits later to completely represent an important, complex section of audio.
• Compression

Representing data in a more efficient format so it takes up less space when stored and/or requires less bandwidth when transmitted.
• Decoding

The process of converting a compressed audio format, like MP3, back into a form that can be played back.
• Digital Audio

Audio that has been sapled to get a number of data points that apporximate the original analog sound waves. Digital audio must be converted back to analog form for playback.
• Digital Audio Extraction (DAE)

Reading digital audio directly from a CD, as opposed to playing it back and sampling the analog signal. DAE is also called 'ripping'.
• Digital Rights Management (DRM)

A system that can track and manage listening and ownership of digital content.
• DMCA

Digital Millenium Copyright Act. Legistation that modified copyright law to make provisions for digital audio, including Webcasting.
• Dolby Surround and Dolby Pro Logic

Dolby Surround is a matrix process that enables any stereo (two-channel) medium, analog or digital, to carry four-channel audio. The encoded audio stream is fully compatible with mono and stereo playback. The four channels are Left, Right, Center, Surround. Dolby Pro Logic enhances Dolby Surround decoder by increasing the separation of Center and Surround channels from Left and Right channels.
• Encoding

The process of transforming audio into a compressed format, such as from WAV to MP3.
• EPAC

Enhanced Perceptual Audio Coder. An audio encoding method developed by Lucent Technologies.
• FLAC

FLAC, an acronym for Free Lossless Audio Codec, is a popular file format for audio data compression. It does not remove any information from the audio stream and is suitable both for everyday playback and audio archival. The FLAC format is currently well supported by many software projects and hardware support is growing.[1] FLAC also supports Replay Gain. Josh Coalson is the primary author of FLAC.
On January 29, 2003, Xiphophorus (now called the Xiph.Org Foundation) announced the incorporation of FLAC into their banner, adding it to Vorbis, Ogg, Theora, Speex, and others.
• ID3 Tag

A means of saving limited descriptive information, including title, artist, album, year of release, genre and comments along with an MP3 track.
• Liquid Audio

A widely used but proprietary digital audio system that uses MPEG-2 AAC.
• Lossy compression

A type of data compression that does not keep all information is said to be 'lossy'. Many popular digital audio formats, including MP3, use lossy compression.
• MP2

MPEG-1 Audio Layer II (MP2, sometimes Musicam) is an audio codec defined by ISO/IEC 11172-3. An extension exists: MPEG-2 Layer II and is defined in ISO/IEC 13818-3. The file extension for files containing such audio data is usually .mp2. While it has largely been superseded by MP3 for PC and Internet applications, it remains a dominant standard for audio broadcasting as part of the DAB digital radio and DVB digital television standards. It is also used internally within the radio industry, for example in NPR's PRSS Content Depot programming distribution system.
• Normalization

The process of adjusting the volume level on tracks as they are encoded, so that their peak levels fall within a similar range.
• OGG

Ogg is a patent-free, fully open and standardised multimedia bitstream container format designed for efficient streaming and manipulation (concatenation and muxing) by the Xiph.Org Foundation.
The name "Ogg" refers to the file format which can be multiplexed with a number of separate independent open source codecs for audio, video and text (e.g. subtitles). Files ending in the .ogg extension may be of any Ogg media filetype, and because the format is free, Ogg's various codecs have been incorporated into a number of different free and commercial media players as well as portable media players from different manufacturers.
• PCM

Pulse Code Modulation. An uncompressed encoding method for digital audio, which is the way audio is stored on music CDs.
• Playlist

A listing of songs quequed up to be played in order. A playlist can also be configured or/and saved for reuse.
• Plug-in

A software module that can be used to add additional functionality to and existing program. The Winamp player uses plug-ins to add special audio or visual effects.
• RealAudio

A popular streaming format for audio and video developed by RealNetworks. The format's use is trageted toward low bitrate streaming.
• RIAA

Recording Industry Association of America. A trade organization that represent music industry financial and legal interests. Among other things, it works to influence legislation, protect the intellctual property rights of artists and record companies, and fight music piracy worldwide.
• Ripping

Is a process of reading the digital information that represents music from the CD. (a slang for digital audio extraction).
• Sampling

The process of repeatedly measuring and storing a digital representation of an analog signal.
• SDMI

Secure Digital Music Initiative is a forum for members of recording, consumer electronics, and computer technology industries to develop a standard framework for secure music distribution and use.
• SHN

Shorten (SHN) is a file format used to losslessly compress CD-quality audio files (44.1 kHz 16-bit stereo PCM). It is a compressed data file format similar to ZIP, RAR, and StuffIt but is optimized for compressing audio data. Lossy formats such as Vorbis and MP3 are more typically used as these are usually ten percent of the size of the original file rather than 50-70 percent, but the smaller file size comes at the cost of data loss (which, depending on the quality of the encoding, the playback equipment, the level of ambient noise during playback, and the listener's hearing, may or may not be perceptible). More mature lossless audio codecs such as FLAC, Monkey's Audio (APE), TTA, and WavPack have become popular recently, although Shorten remains a popular format due to the large number of legally tradable concert recordings in circulation that are encoded as Shorten files. Some applications require (and some hi-fi enthusiasts demand) the lossless digital output that such codecs provide. Shorten files use the .SHN file extension.
The Shorten algorithm and the reference code that implement it were developed by Tony Robinson of Cambridge University in 1992/1993 and later assigned to SoftSound, Ltd. The code was made available under a generous non-commercial license and has since been extended by Wayne Stielau to include seek tables so that one may seek within individual tracks when playing the files on one's computer.
• SmartMedia

A type of small memory card used by many portable MP3 players for storage.
• Sound recording

Legal term for particular recording made of a song by a performer. The copyright for a sound recording is often owned by a recording company.
• Space-shifting

Moving or copying content between different media, formats, or devices.
• Streaming audio

Streaming is a method of transferring multimedia, such as audio or video in one continuous feed from sender or receiver. A streaming server parcels out a little at a time to everyone who is connected to it, and those receiving the stream can play it as it arrives without having to wait for a whole file to download. Streaming audio can be used to create a net radio station.
• TTA

True Audio (abbreviated TTA) is a free, simple real-time lossless audio codec, based on adaptive prognostic filters which has shown satisfactory results comparing to majority of modern analogs.
• VBR

Variable BitRate Encoding attempts to address this shortcoming. It dynamically allocates extra bits during complex sections of the audio track and uses fewer when it can get away with it during simpler sections of the audio. The end result is an MP3 file with a consistent quality level throughout the entire track. The disadvantage is that, it is impossible to predict the size of the finnished MP3 file. As a result, some MP3 players have difficulty with VBR-encoded files because they can't determine the play time of the track.
• VQF (TwinVQ)

The VQF format is the much-shortened name for Transformdomain Weighted Interleave Vector Quantization format, also called TwinVQ. It was developed by Nippon Telegraph and Telephone (NTT) and produces audio files with better compression and better sound quality than MP3. However, compared to MP3, VQF requires substantially greater time for the encoding process. Support for VQF has waned as of late, but MPEG-4 audio will incorporate the technology.
• WAV

WAV (or WAVE), short for Waveform audio format, is a Microsoft and IBM audio file format standard for storing audio on PCs. It is a variant of the RIFF bitstream format method for storing data in "chunks", and thus also close to the IFF and the AIFF format used on Macintosh computers. Both WAVs and AIFFs are compatible with Windows and Macintosh operating systems. It takes into account some differences of the Intel CPU such as little-endian byte order. The RIFF format acts as a "wrapper" for various audio compression codecs. It is the main format used on Windows systems for raw audio.
Though a WAV file can hold compressed audio, the most common WAV format contains uncompressed audio in the pulse-code modulation (PCM) format. PCM audio is the standard audio file format for CDs at 44,100 samples per second, 16 bits per sample. Since PCM uses an uncompressed, lossless storage method, which keeps all the samples of an audio track, professional users or audio experts may use the WAV format for maximum audio quality. WAV audio can also be edited and manipulated with relative ease using software.
• WavPack

WavPack is a free, open source lossless audio compression format developed by David Bryant.
WavPack compression (.WV files) can compress (and restore) 8, 16, 24 & 32-bit float audio files in the .WAV. It also supports multichannel streams and high frequency sampling rates. Like other lossless compression schemes the data reduction varies with the source, but it is generally between 30% and 70% for typical popular music and somewhat better than that for classical music and other sources with greater dynamic range.
WavPack also incorporates a unique "hybrid" mode that provides all the advantages of lossless compression with an additional bonus. Instead of creating a single file, this mode creates both a relatively small, high-quality lossy file that can be used all by itself, and a "correction" file that (when combined with the lossy file) provides full lossless restoration. For some users, this means never having to choose between lossless and lossy compression.
• Windows Media Audio (WMA)

Is part of Microsoft's Windows Media audio and video software, which includes a digital rights management system to secure the files. WMA quality remains quite high even at low bitrates, and Microsoft claims WMA can reproduce CD-quality audio as low as 64kbps, or roughly twice the compression of MP3. WMA is downloadable format, and in conjunction with Microsoft's Advanced Streaming Format (ASF), WMA can also be streamed to listeners. In fact, Windows Media competes directly with the more established RealAudio format and is quickly gaining in popularity. Many portable devices already support WMA, and the format is sure to be a major player in the future of digital audio.

Site References

www.cddb.com
www.doom9.org
www.fhg.de
www.mpeg.org
www.muskalipman.com
www.wikipedia.org
www.xingtech.com
drogo.cslet.it/mpeg/