The bit rate of a digital file how many bits it uses up in a given interval of time. (An audio file is almost always measured in "kilobits per second.") Typically, the higher the bit rate at which music is encoded, the better the sound is. A rate of 128 kilobits/sec is extremely popular in online music downloading because it offers a good compromise between sound quality and download time.
Bit rates between 64 and 128 are great for streaming Internet radio if the listeners have broadband connections. Very low bit rates (below 64kbps are almost totally unsuitable for music but compress voice fairly well and can be used for online voice chat or streaming talk/news radio.
A codec (compression/decompression, coder/decoder, coding/decoding) is an algorithm used for several purposes…
to compress the signal of large files and programs and then to decompress them again when being accessed. Compressing a file usually takes longer than decompressing it.
to transform analog signal (from a microphone, video camera, etc.) into binary ones and zeroes of a digital file. A codec also can be used
to convert data from one digital format to another.
The term "codec" is sometimes used to refer to the computer utilities that do the compression.
There are two broad categories of codecs:
Lossy Codec: A lossy codec is one that discards certain portions of the signal in order to achieve a smaller file size; for example, mp3 codecs attempt to identify and remove portions of the signal that would not result in a perceived loss of quality of the sound to make the file smaller. These losses are sometimes noticeable and sometimes not. The more aggressive the compression setting chosen, the more data is removed. This can result in digital artifacts, which are audible ,errors created by the compression.
Lossless Codec: A lossless codec is one that achieves smaller file sizes through means other than removing data. This can include using a variable bit rate which would use fewer bits to encode silences as compared to an active section of music. Most lossless compression codecs encode repetitive pieces of the signal with symbols and equations that take up less space but provide all the information needed to reconstruct an exact copy of the original.
Some codecs may be configured to be either lossy or lossless. There is often more than one codec for a particular format, and different codecs can vary widely in quality and speed, even for the same format.
This is the ratio between the size of the original uncompressed audio clip and its compressed version. If an audio clip is 20 MB in its native uncompressed form and 1.8MB as a 128k MP3 file, then that file has a compression ration of 11:1 (typical of 128k MP3 files).
DRM is a blanket term for technologies designed to reduce or eliminate digital content piracy. When it comes to audio files, DRM involves the ability to do things like disable playback on foreign or unlicensed machines, limit the transfer of a file to some devices, or prevent it from being burned to multiple CDs. The MP3 file format contains no DRM capabilities, but almost all newer formats do.
A Sample is the set of digital values that characterize a continuous digital signal.
A sampling process with low resolution, with, say, a 2-bit width would give a very limited dynamic range of softest to loudest of only four distinct levels. If more bits are used, more dynamic levels can be measured and stored. A sampling resolution of 16 bits (i.e., 65,536 levels) is often considered adequate for basic sound reproduction. For preservation purposes, a sampling resolution of 24 bits is recommended.
Sampling is the act of converting a continuous signal into discrete measurable units.
The Sampling Rate number of samples per second, measure in Hertz, the unit of frequency (abbreviated to Hz). One Hertz corresponds to one sample per second, and one kilohertz (kHz) corresponds to one thousand samples per second. A sampling rate of 44.1 kHz is often considered adequate for basic sound reproduction. A recommended sampling rate for preservation purposes would be 96 kHz.
AAC is part of the MPEG-2 spec. Technically, the AAC format can support up to 48 full frequency sound channels. It also supports sample rates up to 96 kHz, twice the maximum afforded by MP3. Recently, MPEG-4 AAC added a couple of technologies to the spec that improve quality at extremely low bit rates. At higher bit rates, though, it's essentially the same as MPEG-2 AAC. This is the format used for songs downloaded in the popular iTunes Music Store. Playback is lot supported on all audio players.
AIFF was developed by Apple, for storing high quality music and is appropriate for preservation purposes. It is a uncompressed format and cannot be streamed. AIFF files are usually played without additional plug-ins. It allows specification of sampling rates and sizes. Its files are very large.
AU is a format that most Web browsers man play, but because the files can be larger than other types of audio files, it is best used for short sound clips for effective download times.
BWF is a subset of the more widely used WAVE format, but it also contains extra information about the content. Like WAVE files, BWF files are uncompressed. BWF files are intended for use in radio and television production. The basic audio format is linear PCM, 16 its, sampled at 48 kHz, which is the recommended audio format for production. It may contain MPEG compressed audio data. All WAVE compatible software should be able to play BWF (PCM) audio files. (For more information see "Broadcast Wave Format (BWF) user guide"
http://www.ebu.ch/fr/technical/publications/userguides/bwf_user_guide.php (accessed October 1, 2013)
FLAC is totally free, unpatented, and open-source. It is available for Windows and Unix, with WinAmp and XMMS plugins. As with other lossless compression formats, it produces rather large files, and in fact is not quite as efficient as some other lossless formats. (For more information see "flac (free lossless audio codec)"
http://flac.sourceforge.net/ , accessed October 1, 2013. On how to use it see Audio Key Links: Lossless Compression (PRESTO, Preservation Technologies for European Broadcast Archives)
http://presto.joanneum.ac.at/Public/D4_4.pdf , accessed October 1, 2013.)
MP3 is a compressed audio file format. File sizes vary depending on sampling and bit rate. Can be streamed, but not recommended as it isn't the best format for this — RealAudio and Windows Media are better. Typical compression ratio is 10:1, but it can be as high as 12:1 while maintaining CD sound quality (more less). Samples at 32000, 44100 and 48000 Hz. Files are quite small, but quality is good. The MP3 standard was developed before content protection and online distribution of pirated music became an issue, and thus contains no DRM at all. Because of this, none of the major for-pay online services use MP3 as their file format.
MPEG-4 ALS is an extension to the MPEG-4 audio standard to allow lossless audio compression. It is similar to FLAC in its operation.
MP4 is a multimedia container format standard specified as part of MPEG-4. It is commonly used to store digital audio and digital video streams, especially those defined by MPEG , but can also be used to store streams of other data such as subtitles and still images. Like most modern container formats, MP4 files allows streaming over the Internet. Almost any kind of data can be embedded in MP4 files. The extension .m4p is usually used as the music format for the iTunes Store DRM copy protected content, while .m4a is how being used by the iTunes Store for DRM-free, unprotected music files.
Theora is a free and open video compression format from the Xiph.org Foundation. It can be used to distribute film and video online and on disc without the licensing and royalty fees or vendor lock-in associated with other formats. Theora scales from postage stamp to HD resolution, and is considered particularly competitive at low bitrates. The bitstream format for Theora I was frozen Thursday, 2004 July 1. All bitstreams encoded since that date will remain compatible with future releases.
http://www.theora.org/ (accessed October 1, 2013)
Ogg Vorbis is similar to MP3 or AAC compression formats, but it is free, unpatented, non-proprietary and open-source. "Ogg" is the file container that should one day contain both audio and video, while "Vorbish is the actual audio compression designed to be contained within it. The Vorbis compression scheme is optimized for music and general-purpose audio, not low-bit rate speech compression, and it has no lossless compression option. Vorbis supports 6-channel (5.1) audio, and is fairly well supported by software but almost unseen in the hardware player market. You can find out more about Ogg Vorbis and download the source or tools from the Vorbis Web site - http://www.vorbis.com/ (accessed October 1, 2013).
PCM is a common method of storing and transmitting uncompressed digital audio. Since it is a generic format, it can be read by most audio applications—similar to the way a plain text file can be read by any word-processing program. PCM is used by Audio CDs and digital audio tapes (DATs). PCM is also a very common format for AIFF and WAV files.
QuickTime is a format for encoding and delivering both audio and motion media. Files are stored on a QuickTime server and are "steamed" (gradually delivered on the fly without downloading) to the Web browser. The Server and the Player software are freely available, but the software for creating QuickTime files is a commercial product. Much of the MPEG-4 standard is based on QuickTime, and is widely used for streaming video on the Web. Because the QuickTime format is proprietary, it is not suitable for preservation purposes. (For more on QuickTime see http://www.apple.com/quicktime/ (accessed October 1, 2013 ).)
Real Audio is one of the most common audio formats and was the first widely used system for streaming audio (and video) over the Internet. It yields a very high 10:1 compression ratio. Sound quality is passable, but not high quality. Use it for low-quality delivery files. It provides support for the MP3 format as well. At higher bit rates, it uses MPEG-4 AAC. The new Real 10 platform incorporates RealAudio lossless for true lossless compression and RealAudio Multichannel for up to 5.1 audio, though these formats require RealPlayer 10 or better for playback. Because the Read Audio format is proprietary, it is not suitable for preservation purposes.
WAV is the de-facto standard for basic recording of sounds on Microsoft Windows computers. It is an uncompressed format that produces very high quality sound and created large files. It is an appropriate format for preservation purposes. There are some compressed WAV formats out there, but the most popular is ADPCM. WAV files can usually be played without additional plug-ins. A WAV file can specify an arbitrary sampling rate or 8, 16, or 32 bits. WAV files can be used on both Macs and PCs. It is one of the preferred formats for archiving digital audio. Some archival-quality formats that use a lossless compression are AIFF (Mac), FLAC, MPEG-4/ALS, and MPEG-4/ACC .
Because the WMA format is proprietary, it is not suitable for preservation purposes, but it performs well at low-bit-rates with very good quality at 128 kbps, but not quite up to CD quality.
Windows Media Audio 9: This format supports variable bit rate encoding. You can decode WMA9 files with devices and software made to decode previous generations of WMA.
Windows Media Audio 9 Professional: The Pro edition is similar to WMA9, but supports up to 24-bit/96KHz audio and sound formats up to 5.1 and even 7.1. One of its cool features is that the decoder will automatically adapt the audio material to whatever hardware you have. So if you try to play WMA9 Pro file that is 5.1 at 24-bit/96KHz on sound hardware that can only do stereo 16-bit/48KHz, it will fold the sound down to that spec.
Windows Media Audio 9 Lossless: This is a variable bit rate-only codec that produces absolutely perfect, mathematically lossless copies of an original audio file, including 24-bit/96KHz and 5.1 audio. The compression ratio isn't nearly as high as with lossy compression, averaging from around 2:1 to 4:1 (depending on the complexity of the source material). This codec is designed for professionals and audiophiles that want to archive perfect copies of their music.
Windows Media Audio 9 Voice: This codec is optimized for extremely low bit rate files, like those that you would stream over a dial-up Internet connection or cell phone, or that you would use for real-time online voice chat.
Flash is a proprietary software format for interactive applications on the web, ranging from simple animations to large-scale applications that provide access to data. Macromedia supplies the authoring tool (Flash MX) and the plugin (Flash Player) required for viewing Flash animation and applications in Web browsers. See Adobe Flash CS3 Professional http://www.adobe.com/products/flash.html (accessed October 1, 2013) and Adobe Flash Player http://www.macromedia.com/software/flashplayer/ (accessed October 1, 2013)
AVI files are large, but the quality is very good. They must be encoded/decoded properly by a video plug-in.
MJPEG is an informal name for multimedia formats that compress digital video as a JPEG image. It is often used in mobile appliances such as digital cameras. (For more see "MJPEG" in Wikipedia.)
MJPEG2000 is a codec that stores video as standard JPEG2000 streams. It treats a video stream as a series of still photos, compressing each individually, with no interframe compression. Because it uses no interframe compression, it is ideal for editing. Windows Media Play and ULead Media Studio can use this codec to play, create and edit standard Motion JPEG2000. (A codec is available from Lead Codecs - http://www.leadtools.com/sdk/multimedia/codecs.htm/codecs/lead-jpeg2000.htm, accessed October 1, 2013.)
MPEG is a set of video and audio encoding standards developed by the Moving Picture Experts Group. It can generally produce small files that deliver high quality video through a compression that stores only the changes from one from to another, instead of each entire frame. Of most interest to our repository are the MPEG-1 (Layer 3), MPEG-2 and MPEG-4 formats.
For more information see Moving Picture Experts Group on Wikipedia.org.
MPEG-2 is a video and audio standards for broadcast-quality television and is also used for Digital Versatile Disks (DVD). It offers resolutions of 720 x 480 and 1280 x 720 at 60 fps, with full CD-quality audio.
MPEG-4 is a low bitrate version of MPEG-2 and support for Digital Rights Management . MPEG-4 is a graphics and video compression algorithm standard that is based on MPEG-1 and MPEG-2 and Apple QuickTime technology. Wavelet -based MPEG-4 files are smaller than JPEG or QuickTime files, so they are designed to transmit video and images over a narrower bandwidth and can mix video with text, graphics and 2-D and 3-D animation layers.
MP4 (MPEG-4 Part 14) (.mp4) is a multimedia container format standard specified as part of MPEG-4. It is commonly used to store digital audio and digital video streams, especially those defined by MPEG , but can also be used to store streams of other data such as subtitles and still images. Like most modern container formats, MP4 files allows streaming over the Internet. Almost any kind of data can be embedded in MP4 files. The extension .m4p is usually used as the music format for the iTunes Store DRM copy protected content, while .m4a is now being used by the iTunes Store for DRM-free, unprotected music files.
MPEG-4/ACC (Active Content Compression)
MPEG-4 files using the high-compression, but proprietary Active Content Compression codec.
QuickTime is a format for encoding and delivering motion media and audio files. Files are stored on a QuickTime server and are "streamed" (gradually delivered on the fly without downloading) to the web browser. The Server and the Player software are freely available, but the software for creating QuickTime files is a commercial product. http://www.apple.com/quicktime/
With QTVR software you can ‘stitch" together a series of panoramic still photos and build both a panoramic still image and a complete interactive and navigable panoramic image from the results.
(See QuickTime Virtual Reality - how to make panoramic images, by Brian P. Lawler
http://www.imaging-resource.com/TIPS/LAWLER/PANOHOW2.PDF , accessed May 18, 2007)
RealMedia is a streaming format that uses proprietary encoding. It is equivalent to Windows Media, but requires RealMedia plug-in to be played.
This is a streaming format. Current versions offer near DVD performance.
Digital Audio Tape is often used by professional musicians or reporters to get the highest quality recording. However, because the format uses a rather small tape about 4mm wide, DAT does not hold up to heavy use or archival storage. In order to convert a DAT recording to a computer file, you will need a DAT player, a computer with a compatible sound card and line-level outputs or a TOS-LINK fiber optic cable (for optimal sound quality) to connect the two. For more on DATs, see DAT-Heads .
Digibeta is a digital version of the Betacam format that offers 2:1 compression in a component format. Commonly known as "Digibeta," these tapes are at present (2006) a standard archival remastering format, due to their high quality, durability, and the widespread availability of playback equipment.
DV (Digital Video), originally known as DVC (for Digital Video Cassette) uses 1/4" videotape to record a digital video signal. The acronym DV is also frequently used as a generic term encompassing related compact digital videocassette formats, including MiniDV (which is a DV variant using identical technology in a smaller cassette), DVCAM (which is more robust and made to last longer, but can be play in only good quality Sony players), and DVCPRO. Despite its relatively high quality, the small size of the cassettes and the thinness of the tape mean that DV is not an acceptable archival format.
DVCAM (Digital Video Camera) is Sony's professional variant of the DV standard. It uses the same cassettes as DV and MiniDV, but transports the tape 50% faster. It uses the same codec as regular DV, however, the greater track width lowers the chances of dropout errors. DVCAM tapes (or DV tapes recorded in DVCAM mode) have their recording time reduced by one third. It is still more popular than the more recent DVPRO variety made by Panaeonic..
DVD is a format that shares the same overall dimensions of a CD but has a much greater storage capacity. It uses MPEG2 compression to encode 720 x 480 pixel resolution and full-motion video, and Dolby Digital to encode 5.1 channels of discrete audio. Because of this compression, and because the long-term stability of the discs remains unclear, DVDs are not recommended as an archival video format.
HD-DVD is the disc format for playback of High Definition television signals developed by a group of electronics companies, led by Toshiba. It is a competitor with the Blu-Ray format for control of the market. Because of this conflict, and the accompanying potential for obsolescence, use of HD-DVD should be approached with caution.
Blu-Ray (so named because of the blue-violet laser used in playback) is one of two competing High Definition playback formats currently fighting for dominance. The format was developed by the Blu-ray Disc Association, with the corporate backing of Sony. Because of this conflict, and the accompanying potential for obsolescence, use of Blu-Ray should be approached with caution.
MiniDV (Digital Video) is one of the most commonly used video formats and is aimed at the home consumer market. Its quality is relatively high and the tapes (and equipment) are relatively small. However, that small size does make the cassettes vulnerable to damage and deterioration. MiniDV compresses the video data in order to get more video on the tape. Because of this compression, MiniDV should not be considered a format for long-term storage of video. The image is less stable than on DVCAM.
(Reviewed: September 27, 2008)