Full Rate
Encyclopedia
Full Rate or FR or GSM-FR or GSM 06.10 was the first digital speech coding
standard used in the GSM digital mobile phone system. 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.
) speech coding paradigm. Like many other 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
speech codecs the order is usually 10 and in wideband
speech codecs the order is usually 16.
The speech encoder accepts 13 bit linear PCM at a 8 kHz sample rate.
This can be direct from an analog-to-digital converter
in a phone or computer, or converted from G.711
8-bit nonlinear A-law or μ-law PCM from the PSTN 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.
There is also a Winamp
plugin for raw GSM 06.10 based on the libgsm.
The GSM 06.10 is also used in VoIP software, for example in Ekiga
, QuteCom, Linphone
, Asterisk (PBX)
, Ventrilo
and others.
Speech coding
Speech coding is the 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...
standard used in the GSM digital mobile phone system. 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
Enhanced Full Rate
Enhanced Full Rate or EFR or GSM-EFR or GSM 06.60 is a speech coding standard that was developed in order to improve the quite poor quality of GSM-Full Rate codec. Working at 12.2 kbit/s the EFR provides wirelike quality in any noise free and background noise conditions...
(EFR) and Adaptive Multi-Rate
Adaptive Multi-Rate
The Adaptive Multi-Rate audio codec is a patented audio data compression scheme optimized for speech coding. AMR was adopted as the standard speech codec by 3GPP in October 1999 and is now widely used in GSM and UMTS...
(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 PredictionLong Term Prediction
In GSM, a RPE-LTP scheme is employed in order to reduce the amount of data sent between the mobile station and base transceiver station....
) speech coding paradigm. Like many other speech codecs, linear prediction
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....
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
In radio, narrowband describes a channel in which the bandwidth of the message does not significantly exceed the channel's coherence bandwidth. It is a common misconception that narrowband refers to a channel which occupies only a "small" amount of space on the radio spectrum.The opposite of...
speech codecs the order is usually 10 and in wideband
Wideband
In communications, wideband is a relative term used to describe a wide range of frequencies in a spectrum. A system is typically described as wideband if the message bandwidth significantly exceeds the channel's coherence bandwidth....
speech codecs the order is usually 16.
The speech encoder accepts 13 bit linear PCM at a 8 kHz sample rate.
This can be direct from an analog-to-digital converter
Analog-to-digital converter
An analog-to-digital converter is a device that converts a continuous quantity to a discrete time digital representation. An ADC may also provide an isolated measurement...
in a phone or computer, or converted from G.711
G.711
G.711 is an ITU-T standard for audio companding. It is primarily used in telephony. The standard was released for usage in 1972. Its formal name is Pulse code modulation of voice frequencies. It is required standard in many technologies, for example in H.320 and H.323 specifications. It can also...
8-bit nonlinear A-law or μ-law PCM from the PSTN 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).There is also a Winamp
Winamp
Winamp is a media player for Windows-based PCs and Android devices, written by Nullsoft, now a subsidiary of AOL. It is proprietary freeware/shareware, multi-format, extensible with plug-ins and skins, and is noted for its graphical sound visualization, playlist, and media library features.Winamp...
plugin for raw GSM 06.10 based on the libgsm.
The GSM 06.10 is also used in VoIP software, for example in Ekiga
Ekiga
Ekiga /i k ai g a/ is a VoIP and video conferencing application for GNOME and Windows. It is distributed as free software under the terms of the GNU General Public License. It was the default VoIP client in Ubuntu until October 2009, when it was replaced by Empathy...
, QuteCom, Linphone
Linphone
Linphone is a VoIP application available on PCs running Linux or Windows, Apple computers running Mac OS X, and Android and iPhone mobile phones. It uses the Session Initiation Protocol for communication and is licensed under the GNU General Public License. Linphone uses GTK+ for GUI and on Linux...
, Asterisk (PBX)
Asterisk (PBX)
Asterisk is a software implementation of a telephone private branch exchange ; it was created in 1999 by Mark Spencer of Digium. Like any PBX, it allows attached telephones to make calls to one another, and to connect to other telephone services including the public switched telephone network and...
, Ventrilo
Ventrilo
Ventrilo is a proprietary VoIP software which includes text chat.The Ventrilo client and server are both available as freeware for use with fewer than 8 people on the same server . The server software is available for Microsoft Windows, Mac OS X, or Unix variants such as Linux, Kopi, Solaris,...
and others.
See also
- Half RateHalf RateHalf Rate 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. It is recommended to use this codec when the...
- Enhanced Full RateEnhanced Full RateEnhanced Full Rate or EFR or GSM-EFR or GSM 06.60 is a speech coding standard that was developed in order to improve the quite poor quality of GSM-Full Rate codec. Working at 12.2 kbit/s the EFR provides wirelike quality in any noise free and background noise conditions...
(EFR) - Adaptive Multi-RateAdaptive Multi-RateThe Adaptive Multi-Rate audio codec is a patented audio data compression scheme optimized for speech coding. AMR was adopted as the standard speech codec by 3GPP in October 1999 and is now widely used in GSM and UMTS...
(AMR) - Adaptive Multi-Rate Wideband (AMR-WB)
- Extended Adaptive Multi-Rate - Wideband (AMR-WB+)
- Comparison of audio codecsComparison of audio codecsThe following tables compare general and technical information for a variety of audio formats and audio compression formats. For listening tests comparing the perceived audio quality of audio formats and codecs, see the article Codec listening test....
- RTP audio video profile
External links
- RFC 3551 - RTP payload format for GSM (GSM 06.10)
- ETS 300 961 (GSM 06.10)
- ETS 300 580-2 (GSM 06.10) - legacy specifications