Full Rate (FR), also known as GSM-FR or GSM 06.10 (sometimes simply GSM), was the first digital
speech coding
Speech coding is an application of data compression to 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 family of standards to describe the protocols for second-generation (2G) digital cellular networks, as used by mobile devices such as mobile phones and Mobile broadband modem, mobile broadba ...
digital
mobile phone
A mobile phone or cell phone is a portable telephone that allows users to make and receive calls over a radio frequency link while moving within a designated telephone service area, unlike fixed-location phones ( landline phones). This rad ...
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 is a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates rangi ...
(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 operating in the field of Information and communications technology, information and communications. ETSI supports the de ...
06.10 (ETS 300 961) and is based on RPE-LTP (
Regular Pulse Excitation -
Long Term Prediction) 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 model ...
(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 ...
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 Digital signal (signal processing), digitally represent analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio application ...
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 (signal processing), digi ...
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 the two companding algorithms in the ...
or
μ-law PCM from the
PSTN
The public switched telephone network (PSTN) is the aggregate of the world's telephone networks that are operated by national, regional, or local telephony operators. It provides infrastructure and services for public telephony. The PSTN consists ...
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
(TU Berlin; also known as Berlin Institute of Technology and Technical University of Berlin, although officially the name should not be translated) is a public university, public research university located in Berlin, Germany. It was the first ...
. 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 made from Fermentation in winemaking, fermented fruit. Yeast in winemaking, Yeast consumes the sugar in the fruit and converts it to ethanol and carbon dioxide, releasing heat in the process. Wine is most often made f ...
to provide GSM audio support.
There is also a
Winamp
Winamp is a media player (software), 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 Rad ...
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 known as IP telephony, is a set of technologies used primarily for voice communication sessions over Internet Protocol (IP) networks, such as the Internet. VoIP enables voice calls to be transmitted as ...
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 en ...
,
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 (computing), bandwidth of the Full Rate codec, network capacity for v ...
*
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 is a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates rangi ...
(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
Libgsmhomepage
{{Compression formats
Audio codecs
Speech codecs