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TooLame
TooLAME is a free software MPEG-1 Layer II (MP2) audio encoder written primarily by Mike Cheng. While there are many MP2 encoders, TooLAME is well-known and widely used for its particularly high audio quality. It has been unmaintained since 2003, but is directly succeeded by the TwoLAME code fork (the latest version, TwoLAME 0.4.0, was released October 11, 2019). The name TooLAME is a play on ''LAME'' and ''Layer II''. History After leaving leadership of the LAME project, Mike Cheng decided to redirect his efforts towards the MP2 format. This was in part due to concern with looming legal threats to those distributing software for the widespread MP3 format, due to patents held by Fraunhofer Society, Fraunhofer and Thomson SA, Thomson, while use of MP2 audio was basically unrestricted. For more, see: LAME#Patents and legal issues. The first release of TooLAME (v0.1) was November 7, 1998. He originally based his work on ''mpegaudio.tar''. In October 1999, he started over from s ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Free Software
Free software or libre software is computer software distributed under terms that allow users to run the software for any purpose as well as to study, change, and distribute it and any adapted versions. Free software is a matter of liberty, not price; all users are legally free to do what they want with their copies of a free software (including profiting from them) regardless of how much is paid to obtain the program.Selling Free Software (gnu.org) Computer programs are deemed "free" if they give end-users (not just the developer) ultimate control over the software and, subsequently, over their devices. The right to study and modify a computer program entails that |
Variable Bitrate
Variable bitrate (VBR) is a term used in telecommunications and computing that relates to the bitrate used in sound or video encoding. As opposed to constant bitrate (CBR), VBR files vary the amount of output data per time segment. VBR allows a higher bitrate (and therefore more storage space) to be allocated to the more complex segments of media files while less space is allocated to less complex segments. The average of these rates can be calculated to produce an average bitrate for the file. MP3, WMA and AAC audio files can optionally be encoded in VBR, while Opus and Vorbis are encoded in VBR by default. Variable bit rate encoding is also commonly used on MPEG-2 video, MPEG-4 Part 2 video (Xvid, DivX, etc.), MPEG-4 Part 10/H.264 video, Theora, Dirac and other video compression formats. Additionally, variable rate encoding is inherent in lossless compression schemes such as FLAC and Apple Lossless. Advantages and disadvantages of VBR The advantages of VBR are tha ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
MPEG-1
MPEG-1 is a standard for lossy compression of video and audio. It is designed to compress VHS-quality raw digital video and CD audio down to about 1.5 Mbit/s (26:1 and 6:1 compression ratios respectively) without excessive quality loss, making video CDs, digital cable/satellite TV and digital audio broadcasting (DAB) practical. Today, MPEG-1 has become the most widely compatible lossy audio/video format in the world, and is used in a large number of products and technologies. Perhaps the best-known part of the MPEG-1 standard is the first version of the MP3 audio format it introduced. The MPEG-1 standard is published as ISO/IEC 11172 – Information technology—Coding of moving pictures and associated audio for digital storage media at up to about 1.5 Mbit/s. The standard consists of the following five ''Parts'': #Systems (storage and synchronization of video, audio, and other data together) #Video (compressed video content) #Audio (compressed audio content) #Conformance ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Surround Sound
Surround sound is a technique for enriching the fidelity and depth of sound reproduction by using multiple audio channels from speakers that surround the listener (surround channels). Its first application was in movie theaters. Prior to surround sound, theater sound systems commonly had three ''screen channels'' of sound that played from three loudspeakers (left, center, and right) located in front of the audience. Surround sound adds one or more channels from loudspeakers to the side or behind the listener that are able to create the sensation of sound coming from any horizontal direction (at ground level) around the listener. The technique enhances the perception of sound spatialization by exploiting sound localization: a listener's ability to identify the location or origin of a detected sound in direction and distance. This is achieved by using multiple discrete audio channels routed to an array of loudspeakers. Surround sound typically has a listener location ( sweet ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
MPEG Multichannel
__NOTOC__ MPEG Multichannel is an extension to the MPEG-1 Layer II sound, audio compression specification, as defined in the MPEG-2 Audio standard (International Organization for Standardization, ISO/International Electrotechnical Commission, IEC 13818-3) which allows it provide up to 5.1-channels (surround sound) of audio. To maintain backwards compatibility with the older 2-channel (stereo) audio specification, it uses a channel matrixing scheme, where the additional channels are mixed into the two backwards compatible channels. Extra information in the data stream (ignored by older hardware) contains signals to process extra channels from the matrix. It was originally a mandatory part of the DVD specification for European DVDs, but was dropped in late 1997, and is rarely used as a result. The Super Video CD (SVCD) standard supports MPEG Multichannel. Player support for this audio format is nearly non-existent however, and it is rarely used. MPEG Multichannel audio was propos ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Mencoder
MEncoder is a free command line transcoding tool released under the GNU General Public License. It is a sibling of MPlayer, and can convert all the formats that MPlayer understands into a variety of compressed and uncompressed formats using different codecs. Retrieved on 2009-07-19 MEncoder is included in the MPlayer distribution. Capabilities As it is built from the same code as MPlayer, it can read from every source which MPlayer can read, decode all media which MPlayer can decode and it supports all filters which MPlayer can use. Moreover, MEncoder can read a sequence of and convert it to a video file with ...[...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
MPlayer
MPlayer is a free and open-source media player software application. It is available for Linux, OS X and Microsoft Windows. Versions for OS/2, Syllable, AmigaOS, MorphOS and AROS Research Operating System are also available. A port for DOS using DJGPP is also available. Versions for the Wii Homebrew Channel and Amazon Kindle have also been developed. History Development of MPlayer began in 2000. The original author, Hungarian Árpád Gereöffy, started the project because he was unable to find any satisfactory video players for Linux after XAnim stopped development in 1999. The first version was titled ''mpg12play v0.1'' and was hacked together in a half-hour using ''libmpeg3'' from . After ''mpg12play v0.95pre5'', the code was merged with an AVI player based on ''avifile''s ''Win32 DLL loader'' to form MPlayer v0.3 in November 2000. Gereöffy was soon joined by many other programmers, in the beginning mostly from Hungary, but later worldwide. Alex Beregszászi has maintai ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Digital Audio Broadcasting
Digital radio is the use of digital technology to transmit or receive across the radio spectrum. Digital transmission by radio waves includes digital broadcasting, and especially digital audio radio services. Types In digital broadcasting systems, the analog audio signal is digitized, compressed using an audio coding format such as AAC+ (MDCT) or MP2, and transmitted using a digital modulation scheme. The aim is to increase the number of radio programs in a given spectrum, to improve the audio quality, to eliminate fading problems in mobile environments, to allow additional datacasting services, and to decrease the transmission power or the number of transmitters required to cover a region. However, analog radio (AM and FM) is still more popular and listening to radio over IP (Internet Protocol) is growing in popularity. In 2012 four digital wireless radio systems are recognized by the International Telecommunication Union: the two European systems Digital Audio Broadc ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Broadcast Wave Format
Broadcast Wave Format (BWF) is an extension of the popular Microsoft WAV audio format and is the recording format of most file-based non-linear digital recorders used for motion picture, radio and television production. It was first specified by the European Broadcasting Union in 1997, and updated in 2001 and 2003. It has been accepted as the ITU recommendation ITU-R BS.1352-3, Annex 1. The purpose of this file format is the addition of metadata to facilitate the seamless exchange of sound data between different computer platforms and applications. It specifies the format of metadata, allowing audio processing elements to identify themselves, document their activities, and supports timecode to enable synchronization with other recordings. This metadata is stored as extension chunks in a standard digital audio WAV file. BWF is the recommended format for digitizing sound files by the International Association of Sound and Audiovisual Archives. Files conforming to the Broadca ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Cyclic Redundancy Check
A cyclic redundancy check (CRC) is an error-detecting code commonly used in digital networks and storage devices to detect accidental changes to digital data. Blocks of data entering these systems get a short ''check value'' attached, based on the remainder of a polynomial division of their contents. On retrieval, the calculation is repeated and, in the event the check values do not match, corrective action can be taken against data corruption. CRCs can be used for error correction (see bitfilters). CRCs are so called because the ''check'' (data verification) value is a ''redundancy'' (it expands the message without adding information) and the algorithm is based on ''cyclic'' codes. CRCs are popular because they are simple to implement in binary hardware, easy to analyze mathematically, and particularly good at detecting common errors caused by noise in transmission channels. Because the check value has a fixed length, the function that generates it is occasionally used ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Libsndfile
libsndfile is a widely used C library written by Erik de Castro Lopo for reading and writing audio files. It supports a wide variety of audio file formats and will convert automatically from one to another. It allows the programmer to ignore many details, such as endianness. In addition to the library itself, the package provides command-line programs for converting one format to another (sndfile-convert), for playing audio files (sndfile-play), and for obtaining information about the contents of an audio file (sndfile-info). libsndfile is available for Unix-like systems, including Linux and Mac OS X, and for Microsoft Windows. It is licensed under LGPL-2.1-or-later. libsndfile is used, for example, by audio-editing software such as Audacity and Adobe Audition and the MP3 encoder LAME. See also *Pulse-code modulation * WAV *AIFF Audio Interchange File Format (AIFF) is an audio file format standard used for storing sound data for personal computers and other electro ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
MPEG-1 Layer II
MPEG-1 Audio Layer II or MPEG-2 Audio Layer II (MP2, sometimes incorrectly called Musicam or MUSICAM) is a lossy audio compression format defined by ISO/IEC 11172-3 alongside MPEG-1 Audio Layer I and MPEG-1 Audio Layer III (MP3). While MP3 is much more popular for PC and Internet applications, MP2 remains a dominant standard for audio broadcasting. History of development from MP2 to MP3 MUSICAM MPEG-1 Audio Layer 2 encoding was derived from the MUSICAM (''Masking pattern adapted Universal Subband Integrated Coding And Multiplexing'') audio codec, developed by Centre commun d'études de télévision et télécommunications (CCETT), Philips, and the Institut für Rundfunktechnik (IRT) in 1989 as part of the EUREKA 147 pan-European inter-governmental research and development initiative for the development of a system for the broadcasting of audio and data to fixed, portable or mobile receivers (established in 1987). It began as the Digital Audio Broadcast (DAB) project man ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
