Full Rate (FR or GSM-FR or GSM 06.10 or sometimes simply GSM) was the first digital

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 da ...

standard used in the

GSM The Global System for Mobile Communications (GSM) is a standard developed by the European Telecommunications Standards Institute (ETSI) to describe the protocols for second-generation ( 2G) digital cellular networks used by mobile devices such ...

digital mobile phone system. It uses

linear predictive coding Linear predictive coding (LPC) is a method used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model. ...

(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 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 var ...

(AMR) standards, which provide much higher speech quality with lower bit rate.

Technology

''GSM-FR'' is specified in

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 standard ...

06.10 (ETS 300 961) and is based on RPE-LTP ( Regular Pulse Excitation -

Long Term Prediction In GSM, a Regular Pulse Excitation-Long Term Prediction (RPE-LTP) scheme is employed in order to reduce the amount of data sent between the mobile station (MS) and base transceiver station (BTS). In essence, when a voltage level of a particular spe ...

) speech coding paradigm. Like many other

(LPC) speech codecs,

linear prediction Linear prediction is a mathematical operation where future values of a discrete-time signal are estimated as a linear function of previous samples. In digital signal processing, linear prediction is often called linear predictive coding (LPC) and ...

is used in the synthesis filter. However, unlike most modern speech codecs, the order of the linear prediction is only 8. In modern

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 usua ...

speech codecs the order is usually 10 and in

wideband In communications, a system is wideband when the message bandwidth significantly exceeds the coherence bandwidth of the Channel (communications), channel. Some communication links have such a high Bit rate, data rate that they are forced to use a ...

speech codecs the order is usually 16. The speech encoder accepts 13 bit linear PCM at an 8 kHz sample rate. This can be direct from an

analog-to-digital converter In electronics, an analog-to-digital converter (ADC, A/D, or A-to-D) is a system that converts an analog signal, such as a sound picked up by a microphone or light entering a digital camera, into a digital signal. An ADC may also provide ...

in a phone or computer, or converted from

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 ...

8-bit nonlinear

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 standar ...

or μ-law PCM from the

PSTN The public switched telephone network (PSTN) provides infrastructure and services for public telecommunication. The PSTN is the aggregate of the world's circuit-switched telephone networks that are operated by national, regional, or local teleph ...

with a lookup table. In GSM, the encoded speech is passed to the channel encoder specified in GSM 05.03. In the receive direction, the inverse operations take place. The codec operates on 160 sample frames that span 20 ms, so this is the minimum transcoder delay possible even with infinitely fast CPUs and zero network latency. The operational requirement is that the transcoder delay should be less than 30 ms. The transcoder delay is defined as the time interval between the instant a speech frame of 160 samples has been received at the encoder input and the instant the corresponding 160 reconstructed speech samples have been out-put by the speech decoder at an 8 kHz sample rate.

Implementations

The free ''libgsm'' codec can encode and decode GSM Full Rate audio. "libgsm" was developed 1992–1994 by Jutta Degener and Carsten Bormann, then at

Technische Universität Berlin The Technical University of Berlin (official name both in English and german: link=no, Technische Universität Berlin, also known as TU Berlin and Berlin Institute of Technology) is a public research university located in Berlin, Germany. It was ...

. Since a GSM speech frame is 32.5 bytes, this implementation also defined a 33-byte nibble-padded representation of a GSM frame (which, at a frame rate of 50/s, is the basis for the incorrect claim that the GSM bit rate is 13.2 kbit/s). This codec can also be compiled into

Wine Wine is an alcoholic drink typically made from fermented grapes. Yeast consumes the sugar in the grapes and converts it to ethanol and carbon dioxide, releasing heat in the process. Different varieties of grapes and strains of yeasts are m ...

to provide GSM audio support. There is also a

Winamp Winamp is a media player for Microsoft Windows originally developed by Justin Frankel and Dmitry Boldyrev by their company Nullsoft, which they later sold to AOL in 1999 for $80 million. It was then acquired by Radionomy in 2014. Sinc ...

plugin for raw GSM 06.10 based on the libgsm.Cedric Hans (2004-06-08
Winamp Plug-in - Raw GSM Winamp Plugin
, Winamp, Retrieved 2009-10-09 The GSM 06.10 is also used in

VoIP Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet t ...

software, for example in

Ekiga Ekiga (formerly called GnomeMeeting) is a VoIP and video conferencing application for GNOME and Microsoft Windows. It is distributed as free software under the terms of the GNU GPL-2.0-or-later. It was the default VoIP client in Ubuntu until Octob ...

QuteCom QuteCom (previously called WengoPhone) was a free-software SIP-compliant VoIP client developed by the QuteCom (previously OpenWengo) community under the GPL-2.0-or-later license. It allows users to speak to other users of SIP-compliant VoIP sof ...

Linphone __NOTOC__ Linphone (contraction of ''Linux phone'') is a free voice over IP softphone, SIP client and service. It may be used for audio and video direct calls and calls through any VoIP softswitch or IP-PBX. Linphone also provides the possibilit ...

Asterisk (PBX) Asterisk is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication ...

Ventrilo Ventrilo (or Vent for short) is a proprietary VoIP software that includes text chat. The Ventrilo client and server are both available as freeware for use with up to 8 people on the same server. Rented servers can maintain up to 400 people. T ...

and others.

Technology

Implementations

See also

References

External links