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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 d ...
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 mod ...
(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 sp ...
) speech coding paradigm. Like many other
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 mod ...
(LPC) speech codecs, linear prediction 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. Some communication links have such a high data rate that they are forced to use a wide bandwidth; other links ma ...
speech codecs the order is usually 16. The speech encoder accepts 13 bit linear
PCM Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the am ...
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 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. 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 Fermentation in winemaking, fermented grapes. Yeast in winemaking, Yeast consumes the sugar in the grapes and converts it to ethanol and carbon dioxide, releasing heat in the process. Different ...
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. Since ve ...
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, QuteCom, Linphone,
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 e ...
, Ventrilo and others.


See also

*
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 ...
* Enhanced Full Rate (EFR) *
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) * Adaptive Multi-Rate Wideband (AMR-WB) * Extended Adaptive Multi-Rate - Wideband (AMR-WB+) *
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 informati ...
* RTP audio video profile


References


External links


RFC 3551
- RTP payload format for GSM (GSM 06.10)
ETS 300 961 (GSM 06.10)
- European Standard
ETS 300 580-2 (GSM 06.10)
- legacy specifications

- Technical Specification
Libgsm
homepage {{Compression formats Audio codecs Speech codecs