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G.711
G.711 is a narrowband audio codec originally designed for use in telephony that provides toll-quality audio at 64 kbit/s. G.711 passes audio signals in the range of 300–3400 Hz and samples them at the rate of 8,000 samples per second, with the tolerance on that rate of 50 parts per million (ppm). Non-uniform (logarithmic) quantization with 8 bits is used to represent each sample, resulting in a 64 kbit/s bit rate. There are two slightly different versions: μ-law, which is used primarily in North America and Japan, and A-law, which is in use in most other countries outside North America. G.711 is an ITU-T standard (Recommendation) for audio companding, titled Pulse code modulation (PCM) of voice frequencies released for use in 1972. It is a required standard in many technologies, such as in the H.320 and H.323 standards. It can also be used for fax communication over IP networks (as defined in T.38 specification). Two enhancements to G.711 have been publishe ... [...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] |
Comparison Of Audio Coding Formats
The following tables compare general and technical information for a variety of audio coding formats. For listening tests comparing the perceived audio quality of audio formats and codecs, see the article Codec listening test. General information Notes # The 'Music' category is merely a guideline on commercialized uses of a particular format, not a technical assessment of its capabilities. (For example, in terms of marketshare, MP3 and AAC dominate the personal audio market, though many other formats are comparably well suited to fill this role from a purely technical standpoint.) # First public release date is first of either specification publishing or source releasing, or in the case of closed-specification, closed-source codecs, is the date of first binary releasing. Many developing codecs have pre-releases consisting of pre-1.0 versions and perhaps 1.0 release candidates (RCs), although 1.0 may not necessarily be the release version. # Latest stable version is that o ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
μ-law Algorithm
The μ-law algorithm (sometimes written mu-law, often approximated as u-law) is a companding algorithm, primarily used in 8-bit PCM digital telecommunication systems in North America and Japan. It is one of two versions of the G.711 standard from ITU-T, the other version being the similar A-law. A-law is used in regions where digital telecommunication signals are carried on E-1 circuits, e.g. Europe. Companding algorithms reduce the dynamic range of an audio signal. In analog systems, this can increase the signal-to-noise ratio (SNR) achieved during transmission; in the digital domain, it can reduce the quantization error (hence increasing the signal-to-quantization-noise ratio). These SNR increases can be traded instead for reduced bandwidth for equivalent SNR. Algorithm types The μ-law algorithm may be described in an analog form and in a quantized digital form. Continuous For a given input , the equation for μ-law encoding is: : F(x) = \sgn(x) \dfrac, \quad -1 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
A-law
An A-law algorithm is a standard companding algorithm, used in European 8-bit PCM digital communications systems to optimize, i.e. modify, the dynamic range of an analog signal for digitizing. It is one of two versions of the G.711 standard from ITU-T, the other version being the similar μ-law, used in North America and Japan. For a given input x, the equation for A-law encoding is as follows: F(x) = \sgn(x) \begin \dfrac, & , x, < \dfrac, \\ ex \dfrac, & \dfrac \leq , x, \leq 1, \end where is the compression parameter. In Europe, . A-law expansion is given by the inverse function: The reason for this encoding is that the wide dyn ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
A-law Algorithm
An A-law algorithm is a standard companding algorithm, used in European 8-bit PCM digital communications systems to optimize, i.e. modify, the dynamic range of an analog signal for digitizing. It is one of two versions of the G.711 standard from ITU-T, the other version being the similar μ-law, used in North America and Japan. For a given input x, the equation for A-law encoding is as follows: F(x) = \sgn(x) \begin \dfrac, & , x, < \dfrac, \\ ex \dfrac, & \dfrac \leq , x, \leq 1, \end where is the compression parameter. In Europe, . A-law expansion is given by the inverse function: The reason for this encoding is that the wide dyn ... [...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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Packet Loss Concealment
Packet loss concealment (PLC) is a technique to mask the effects of packet loss in voice over IP (VoIP) communications. When the voice signal is sent as VoIP packets on an IP network, the packets may (and likely will) travel different routes. A packet therefore might arrive very late, might be corrupted, or simply might not arrive at all. One example case of the last situation could be, when a packet is rejected by a server which has a full buffer and cannot accept any more data. Other cases include network congestion resulting in significant delay. In a VoIP connection, error-control techniques such as automatic repeat request (ARQ) are not feasible and the receiver should be able to cope with packet loss. Packet loss concealment is the inclusion in a design of methodologies for accounting for and compensating for the loss of voice packets. PLC techniques * Zero insertion: the lost speech frames are replaced with silence. * Waveform substitution: the missing gap is reconstruct ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Two's Complement
Two's complement is a mathematical operation to reversibly convert a positive binary number into a negative binary number with equivalent (but negative) value, using the binary digit with the greatest place value (the leftmost bit in big- endian numbers, rightmost bit in little-endian numbers) to indicate whether the binary number is positive or negative (the sign). It is used in computer science as the most common method of representing signed (positive, negative, and zero) integers on computers, and more generally, fixed point binary values. When the most significant bit is a one, the number is signed as negative. . Two's complement is executed by 1) inverting (i.e. flipping) all bits, then 2) adding a place value of 1 to the inverted number. For example, say the number −6 is of interest. +6 in binary is 0110 (the leftmost most significant bit is needed for the sign; positive 6 is not 110 because it would be interpreted as -2). Step one is to flip all bits, yielding 1001. ... [...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] |
MDCT
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] |
Wideband
In communications, a system is wideband when the message bandwidth significantly exceeds the coherence bandwidth of the channel. Some communication links have such a high data rate that they are forced to use a wide bandwidth; other links may have relatively low data rates, but deliberately use a wider bandwidth than "necessary" for that data rate in order to gain other advantages; see '' spread spectrum''. A wideband antenna is one with approximately or exactly the same operating characteristics over a very wide Passband. It is distinguished from broadband antennas, where the passband is large, but the antenna gain and/or radiation pattern need not stay the same over the passband. The term Wideband Audio or (also termed HD Voice or Wideband Voice) denotes a telephony using a wideband codec, which uses a greater frequency range of the audio spectrum than conventional voiceband telephone calls, resulting in a clearer sound. Wideband in this context is usually considered ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Clock Recovery
In serial communication of digital data, clock recovery is the process of extracting timing information from a serial data stream itself, allowing the timing of the data in the stream to be accurately determined without separate clock information. It is widely used in data communications; the similar concept used in analog systems like color television is known as carrier recovery. Basic concept Serial data is normally sent as a series of pulses with well-defined timing constraints. This presents a problem for the receiving side; if their own local clock is not precisely synchronized with the transmitter, they may sample the signal at the wrong time and thereby decode the signal incorrectly. This can be addressed with extremely accurate and stable clocks, like atomic clocks, but these are expensive and complex. More common low-cost clock systems, like quartz oscillators, are accurate enough for this task over short periods of time, but over a period of minutes or hours the drift ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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