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Half Rate
Half Rate (HR or GSM-HR or GSM 06.20) is a speech coding system for GSM, developed in the early 1990s. Since the codec, operating at 5.6 kbit/s, requires half the bandwidth of the Full Rate codec, network capacity for voice traffic is doubled, at the expense of audio quality. The sampling rate is 8 kHz with resolution 13 bit, frame length 160 samples (20 ms) and subframe length 40 samples (5 ms). GSM Half Rate is specified in ETSI EN 300 969 (GSM 06.20), and uses a form of the VSELP algorithm. Previous specification was in ETSI ETS 300 581–2, which first edition was published in December 1995. Retrieved on 2009-07-11 For some Nokia phones one can configure the use of this codec: * To a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Speech Coding
Speech coding is an application of data compression of digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream. Some applications of speech coding are mobile telephony and voice over IP (VoIP). The most widely used speech coding technique in mobile telephony is linear predictive coding (LPC), while the most widely used in VoIP applications are the LPC and modified discrete cosine transform (MDCT) techniques. The techniques employed in speech coding are similar to those used in audio data compression and audio coding where knowledge in psychoacoustics is used to transmit only data that is relevant to the human auditory system. For example, in voiceband speech coding, only information in the frequency band 400 to 3500 Hz is transmitted but the reconst ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Codec
A codec is a device or computer program that encodes or decodes a data stream or signal. ''Codec'' is a portmanteau of coder/decoder. In electronic communications, an endec is a device that acts as both an encoder and a decoder on a signal or data stream, and hence is a type of codec. ''Endec'' is a portmanteau of encoder/decoder. A coder or encoder encodes a data stream or a signal for transmission or storage, possibly in encrypted form, and the decoder function reverses the encoding for playback or editing. Codecs are used in videoconferencing, streaming media, and video editing applications. History In the mid-20th century, a codec was a device that coded analog signals into digital form using pulse-code modulation (PCM). Later, the name was also applied to software for converting between digital signal formats, including companding functions. Examples An audio codec converts analog audio signals into digital signals for transmission or encodes them for storage. A receiv ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Bandwidth (computing)
In computing, bandwidth is the maximum rate of data transfer across a given path. Bandwidth may be characterized as network bandwidth, data bandwidth, or digital bandwidth. This definition of ''bandwidth'' is in contrast to the field of signal processing, wireless communications, modem data transmission, digital communications, and electronics, in which ''bandwidth'' is used to refer to analog signal bandwidth measured in hertz, meaning the frequency range between lowest and highest attainable frequency while meeting a well-defined impairment level in signal power. The actual bit rate that can be achieved depends not only on the signal bandwidth but also on the noise on the channel. Network capacity The term ''bandwidth'' sometimes defines the net bit rate 'peak bit rate', 'information rate,' or physical layer 'useful bit rate', channel capacity, or the maximum throughput of a logical or physical communication path in a digital communication system. For example, bandwidth test ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Full Rate
Full Rate (FR or GSM-FR or GSM 06.10 or sometimes simply GSM) was the first digital speech coding standard used in the GSM digital mobile phone system. It uses linear predictive coding (LPC). The bit rate of the codec is 13 kbit/s, or 1.625 bits/audio sample (often padded out to 33 bytes/20 ms or 13.2 kbit/s). The quality of the coded speech is quite poor by modern standards, but at the time of development (early 1990s) it was a good compromise between computational complexity and quality, requiring only on the order of a million additions and multiplications per second. The codec is still widely used in networks around the world. Gradually FR will be replaced by Enhanced Full Rate (EFR) and Adaptive Multi-Rate (AMR) standards, which provide much higher speech quality with lower bit rate. Technology ''GSM-FR'' is specified in ETSI 06.10 (ETS 300 961) and is based on RPE-LTP ( Regular Pulse Excitation - Long Term Prediction) speech coding paradigm. Like many other linear predicti ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
European Telecommunications Standards Institute
The European Telecommunications Standards Institute (ETSI) is an independent, not-for-profit, standardization organization in the field of Information and communications technology, information and communications. ETSI supports the development and testing of global technical standards for ICT-enabled systems, applications and services. Overview ETSI was set up in 1988 by the European Conference of Postal and Telecommunications Administrations (European Conference of Postal and Telecommunications Administrations, CEPT) following a proposal from the European Commission. ETSI is the officially recognized body with a responsibility for the standardization of Information and Communication Technologies (ICT). It is one of the three bodies, the others being European Committee for Standardization, CEN and CENELEC, officially recognized by the European Union as a European Standards Organization (ESO). The role of the European Standards Organizations is to support EU regulation and poli ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Vector Sum Excited Linear Prediction
Vector sum excited linear prediction (VSELP) is a speech coding method used in several cellular standards. The VSELP algorithm is an analysis-by-synthesis coding technique and belongs to the class of speech coding algorithms known as CELP (Code Excited Linear Prediction).ETSIEN 300 969 - Half rate speech transcoding (GSM 06.20 version 8.0.1 Release 1999) Retrieved on 2009-07-11 Variations of this codec have been used in several 2G cellular telephony standards, including IS-54, IS-136 (D-AMPS), GSM (Half Rate speech) and iDEN. It was also used in the first version of RealAudio for audio over the Internet. The IS-54 VSELP standard was published by the Telecommunications Industry Association in 1989. D-AMPS (IS-54 and IS-136) VSELP specifies an encoding of each 20 ms of speech into 159-bit frames, thus achieving a raw data rate of 7.95 kbit/s. In an actual TDMA cell phone, the vocoder output is packaged with error correction and signaling information, resulting in an over-the-air da ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
ETSI
The European Telecommunications Standards Institute (ETSI) is an independent, not-for-profit, standardization organization in the field of information and communications. ETSI supports the development and testing of global technical standards for ICT-enabled systems, applications and services. Overview ETSI was set up in 1988 by the European Conference of Postal and Telecommunications Administrations ( CEPT) following a proposal from the European Commission. ETSI is the officially recognized body with a responsibility for the standardization of Information and Communication Technologies (ICT). It is one of the three bodies, the others being CEN and CENELEC, officially recognized by the European Union as a European Standards Organization (ESO). The role of the European Standards Organizations is to support EU regulation and policies through the production of Harmonised European Standards and other deliverables. The standards developed by ESOs are the only ones that can be rec ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Adaptive Multi-Rate
The Adaptive Multi-Rate (AMR, AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding. AMR speech codec consists of a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality speech starting at 7.4 kbit/s. AMR was adopted as the standard speech codec by 3GPP in October 1999 and is now widely used in GSM and UMTS. It uses link adaptation to select from one of eight different bit rates based on link conditions. AMR is also a file format for storing spoken audio using the AMR codec. Many modern mobile telephone handsets can store short audio recordings in the AMR format, and both Free software, free and proprietary programs exist (see #Software support, Software support) to convert between this and other formats, although AMR is a speech format and is unlikely to give ideal results for other audio. The common filename extension is .amr. ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Adaptive Multi-Rate Wideband
Adaptive Multi-Rate Wideband (AMR-WB) is a patented Wideband audio, wideband speech coding, speech audio coding standard developed based on Adaptive Multi-Rate audio codec, Adaptive Multi-Rate encoding, using a similar methodology to algebraic code-excited linear prediction (ACELP). AMR-WB provides improved speech quality due to a wider speech bandwidth of 50–7000 Hz compared to narrowband speech coders which in general are optimized for Plain old telephone service, POTS wireline quality of 300–3400 Hz. AMR-WB was developed by Nokia and VoiceAge and it was first specified by 3GPP. AMR-WB is codified as G.722.2, an ITU-T standard speech codec, formally known as ''Wideband coding of speech at around 16 kbit/s using Adaptive Multi-Rate Wideband (AMR-WB)''. G.722.2 AMR-WB is the same codec as the 3GPP AMR-WB. The corresponding 3GPP specifications are TS 26.190 for the speech codec and TS 26.194 for the Voice Activity Detector. The AMR-WB format has the following paramete ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Extended Adaptive Multi-Rate - Wideband
Extension, extend or extended may refer to: Mathematics Logic or set theory * Axiom of extensionality * Extensible cardinal * Extension (model theory) * Extension (predicate logic), the set of tuples of values that satisfy the predicate * Extension (semantics), the set of things to which a property applies * Extension by definitions * Extensional definition, a definition that enumerates every individual a term applies to * Extensionality Other uses * Extension of a polyhedron, in geometry * Exterior algebra, Grassmann's theory of extension, in geometry * Homotopy extension property, in topology * Kolmogorov extension theorem, in probability theory * Linear extension, in order theory * Sheaf extension, in algebraic geometry * Tietze extension theorem, in topology * Whitney extension theorem, in differential geometry * Group extension, in abstract algebra and homological algebra Music * Extension (music), notes that fit outside the standard range * ''Extended'' (Solar Fields ... [...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] |
