Full Rate or FR or GSM-FR or GSM 06.10 was the first digital speech coding standard used in GSM digital mobile phone system. The average bit rate of the codec is 13 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. 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.

Contents 1 Technology 2 Implementations 3 See also 4 References 5 External links

[edit] 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 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 takes its input as a 13 bit uniform PCM signal either from the audio part of the mobile station or on the network side, from the PSTN via an 8 bit A-law or μ-law (PCS 1900) to 13 bit uniform PCM conversion. The encoded speech at the output of the speech encoder is delivered to a channel encoder unit which is specified in GSM 05.03. In the receive direction, the inverse operations take place.

The sampling rate is 8000 sample/s. The theoretical minimum transcoder delay which can be achieved is 20 ms. The 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.^[1]

[edit] Implementations

The free libgsm codec can encode and decode GSM Full Rate audio.^[2][3] "libgsm" was developed by Jutta Degener and Carsten Bormann in Technical University of Berlin.^[4]

There is also a Winamp plugin for raw GSM 06.10 based on the libgsm.^[5][6]

The GSM 06.10 is also used in VoIP software, for example in Ekiga, QuteCom, Linphone, Asterisk (PBX) and others.

[edit] See also

• Half Rate
• Enhanced Full Rate (EFR)
• Adaptive Multi-Rate (AMR)
• Adaptive Multi-Rate Wideband (AMR-WB)
• Extended Adaptive Multi-Rate - Wideband (AMR-WB+)
• Comparison of audio codecs
• RTP audio video profile

[edit] References

[edit] 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