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G.729.1
G.729.1 is an 8-32 kbit/s embedded speech and audio codec providing bitstream interoperability with G.729, G.729 Annex A and G.729 Annex B. Its official name is ''G.729-based embedded variable bit rate codec: An 8-32 kbit/s scalable wideband coder bitstream interoperable with G.729''. It was introduced in 2006. This codec has been designed to provide better quality and more flexibility than the existing ITU-T G.729 speech coding standard. G.729.1 is scalable in bit rate, acoustic bandwidth and complexity. In addition it offers various encoder and decoder modes, including the support of both 8 and 16 kHz input/output sampling frequency, compatibility with G.729B, and reduced algorithmic delay. The bitstream of G.729.1 is structured into 12 hierarchical layers. The first layer (or core layer) at 8 kbit/s follows the G.729 format. The second layer (adds 4 kbit/s for a total of 12 kbit/s) is a narrowband enhancement layer. The third layer (2 kbit/s for a total of 14 kbit/s) i ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
List Of Codecs
The following is a list of compression formats and related codecs. Audio compression formats Non-compression * Linear pulse-code modulation (LPCM, generally only described as PCM) is the format for uncompressed audio in media files and it is also the standard for CD-DA; note that in computers, LPCM is usually stored in container formats such as WAV, AIFF, or AU, or as raw audio format, although not technically necessary. ** FFmpeg * Pulse-density modulation (PDM) ** Direct Stream Digital (DSD) is standard for Super Audio CD *** foobar2000 Super Audio CD Decoder (based on MPEG-4 DST reference decoder) *** FFmpeg (based on dsd2pcm) * Pulse-amplitude modulation (PAM) Lossless compression * Actively used ** Most popular *** Free Lossless Audio Codec (FLAC) **** libFLAC **** FFmpeg *** Apple Lossless Audio Codec (ALAC) **** Apple QuickTime **** libalac **** FFmpeg **** Apple Music *** Monkey's Audio (APE) **** Monkey's Audio SDK **** FFmpeg (decoder only) *** OptimFROG ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Modified Discrete Cosine Transform
The modified discrete cosine transform (MDCT) is a transform based on the type-IV discrete cosine transform (DCT-IV), with the additional property of being lapped: it is designed to be performed on consecutive blocks of a larger dataset, where subsequent blocks are overlapped so that the last half of one block coincides with the first half of the next block. This overlapping, in addition to the energy-compaction qualities of the DCT, makes the MDCT especially attractive for signal compression applications, since it helps to avoid artifacts stemming from the block boundaries. As a result of these advantages, the MDCT is the most widely used lossy compression technique in audio data compression. It is employed in most modern audio coding standards, including MP3, Dolby Digital (AC-3), Vorbis (Ogg), Windows Media Audio (WMA), ATRAC, Cook, Advanced Audio Coding (AAC), High-Definition Coding (HDC), LDAC, Dolby AC-4, and MPEG-H 3D Audio, as well as speech coding standards suc ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Time-domain Aliasing Cancellation
The modified discrete cosine transform (MDCT) is a transform based on the type-IV discrete cosine transform (DCT-IV), with the additional property of being lapped: it is designed to be performed on consecutive blocks of a larger dataset, where subsequent blocks are overlapped so that the last half of one block coincides with the first half of the next block. This overlapping, in addition to the energy-compaction qualities of the DCT, makes the MDCT especially attractive for signal compression applications, since it helps to avoid artifacts stemming from the block boundaries. As a result of these advantages, the MDCT is the most widely used lossy compression technique in audio data compression. It is employed in most modern audio coding standards, including MP3, Dolby Digital (AC-3), Vorbis (Ogg), Windows Media Audio (WMA), ATRAC, Cook, Advanced Audio Coding (AAC), High-Definition Coding (HDC), LDAC, Dolby AC-4, and MPEG-H 3D Audio, as well as speech coding standards such as ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Code-excited Linear Prediction
Code-excited linear prediction (CELP) is a linear predictive speech coding algorithm originally proposed by Manfred R. Schroeder and Bishnu S. Atal in 1985. At the time, it provided significantly better quality than existing low bit-rate algorithms, such as residual-excited linear prediction (RELP) and linear predictive coding (LPC) vocoders (e.g., FS-1015). Along with its variants, such as algebraic CELP, relaxed CELP, low-delay CELP and vector sum excited linear prediction, it is currently the most widely used speech coding algorithm. It is also used in MPEG-4 Audio speech coding. CELP is commonly used as a generic term for a class of algorithms and not for a particular codec. Background The CELP algorithm is based on four main ideas: * Using the source-filter model of speech production through linear prediction (LP) (see the textbook "speech coding algorithm"); * Using an adaptive and a fixed codebook as the input (excitation) of the LP model; * Performing a search in ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
ITU-T
The ITU Telecommunication Standardization Sector (ITU-T) is one of the three sectors (divisions or units) of the International Telecommunication Union (ITU). It is responsible for coordinating standards for telecommunications and Information Communication Technology such as X.509 for cybersecurity, Y.3172 and Y.3173 for machine learning, and H.264/MPEG-4 AVC for video compression, between its Member States, Private Sector Members, and Academia Members. The first meeting of the World Telecommunication Standardization Assembly (WTSA), the sector's governing conference, took place on 1 March of that year. ITU-T has a permanent secretariat called the Telecommunication Standardization Bureau (TSB), which is based at the ITU headquarters in Geneva, Switzerland. The current director of the TSB is Chaesub Lee (of South Korea), whose first 4-year term commenced on 1 January 2015, and whose second 4-year term commenced on 1 January 2019. Chaesub Lee succeeded Malcolm Johnson of the Uni ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Narrowband
Narrowband signals are signals that occupy a narrow range of frequencies or that have a small fractional bandwidth. In the audio spectrum, narrowband sounds are sounds that occupy a narrow range of frequencies. In telephony, narrowband is usually considered to cover frequencies 300–3400 Hz, i.e. the voiceband. In radio communications, a narrowband channel is a channel in which the bandwidth of the message does not significantly exceed the channel's coherence bandwidth. In the study of wired channels, ''narrowband'' implies that the channel under consideration is sufficiently narrow that its frequency response can be considered flat. The message bandwidth will therefore be less than the coherence bandwidth of the channel. That is, no channel has perfectly flat fading, but the analysis of many aspects of wireless systems is greatly simplified if flat fading can be assumed. Two-way radio narrowband Two-Way Radio Narrowbanding refers to a U.S. Federal Communications Com ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Speech Codecs
Speech is a human vocal communication using language. Each language uses phonetic combinations of vowel and consonant sounds that form the sound of its words (that is, all English words sound different from all French words, even if they are the same word, e.g., "role" or "hotel"), and using those words in their semantic character as words in the lexicon of a language according to the syntactic constraints that govern lexical words' function in a sentence. In speaking, speakers perform many different intentional speech acts, e.g., informing, declaring, asking, persuading, directing, and can use enunciation, intonation, degrees of loudness, tempo, and other non-representational or paralinguistic aspects of vocalization to convey meaning. In their speech, speakers also unintentionally communicate many aspects of their social position such as sex, age, place of origin (through accent), physical states (alertness and sleepiness, vigor or weakness, health or illness), psychologica ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Codecs
An audio codec is a device or computer program capable of encoding or decoding a digital data stream (a codec) that encodes or decodes audio. In software, an audio codec is a computer program implementing an algorithm that compresses and decompresses digital audio data according to a given audio file or streaming media audio coding format. The objective of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining quality. This can effectively reduce the storage space and the bandwidth required for transmission of the stored audio file. Most software codecs are implemented as libraries which interface to one or more multimedia players. Most modern audio compression algorithms are based on modified discrete cosine transform (MDCT) coding and linear predictive coding (LPC). In hardware, audio codec refers to a single device that encodes analog audio as digital signals and decodes digital back into analog. In other words, it contains bot ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Royalty-free
Royalty-free (RF) material subject to copyright or other intellectual property rights may be used without the need to pay royalties or license fees for each use, per each copy or volume sold or some time period of use or sales. Computer standards Many computer industry standards, especially those developed and submitted by industry consortiums or individual companies, involve royalties for the actual implementation of these standards. These royalties are typically charged on a "per port"/"per device" basis, where the manufacturer of end-user devices has to pay a small fixed fee for each device sold, and also include a substantial annual fixed fee. With millions of devices sold each year, the royalties can amount to several millions of dollars, which is a significant burden for the manufacturer. Examples of such royalties-based standards include IEEE 1394, HDMI, and H.264/ MPEG-4 AVC. Royalty-free standards do not include any "per-port" or "per-volume" charges or annual payments ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Software Patent
A software patent is a patent on a piece of software, such as a computer program, libraries, user interface, or algorithm. Background A patent is a set of exclusionary rights granted by a state to a patent holder for a limited period of time, usually 20 years. These rights are granted to patent applicants in exchange for their disclosure of the inventions. Once a patent is granted in a given country, no person may make, use, sell or import/export the claimed invention in that country without the permission of the patent holder. Permission, where granted, is typically in the form of a license which conditions are set by the patent owner: it may be free or in return for a royalty payment or lump sum fee. Patents are territorial in nature. To obtain a patent, inventors must file patent applications in each and every country in which they want a patent. For example, separate applications must be filed in Japan, China, the United States and India if the applicant wishes to obtain ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Transform Coding
Transform coding is a type of data compression for "natural" data like audio signals or photographic images. The transformation is typically lossless (perfectly reversible) on its own but is used to enable better (more targeted) quantization, which then results in a lower quality copy of the original input (lossy compression). In transform coding, knowledge of the application is used to choose information to discard, thereby lowering its bandwidth. The remaining information can then be compressed via a variety of methods. When the output is decoded, the result may not be identical to the original input, but is expected to be close enough for the purpose of the application. Colour television NTSC One of the most successful transform encoding system is typically not referred to as such—the example being NTSC color television. After an extensive series of studies in the 1950s, Alda Bedford showed that the human eye has high resolution only for black and white, somewhat le ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Kilobit
The kilobit is a multiple of the unit bit for digital information or computer storage. The prefix '' kilo-'' (symbol k) is defined in the International System of Units (SI) as a multiplier of 103 (1 thousand), and therefore, :1 kilobit = = 1000 bits. The kilobit has the unit symbol kbit or kb. Using the common byte size of 8 bits, 1 kbit is equal to 125 bytes. The kilobit is commonly used in the expression of data rates of digital communication circuits as kilobits per second (kbit/s or kb/s), or abbreviated as ''kbps'', as in, for example, ''a 56 kbps PSTN circuit'', or ''a 512 kbit/s broadband Internet connection''. The unit symbol kb (lowercase 'b') is typographically similar to the international standard unit symbol for the kilobyte, i.e. kB (upper case 'B'). The International Electrotechnical Commission (IEC) recommends the symbol bit instead of b. The prefix ''kilo-'' is often used in fields of computer science and information technology ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
