Tcat Installation On A Debian, Ubuntu, Alpine, Arch, Kali, CentOS, Fedora And Raspbian
tcat
User binaries for a GSM speech compressor
Install tcat
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Debian
apt-get install libgsm-toolsClick to copy -
Ubuntu
apt-get install libgsm-toolsClick to copy -
Alpine OS
apk add gsmClick to copy -
Arch Linux
pacman -S gsmClick to copy -
Kali Linux
apt-get install libgsm-toolsClick to copy -
CentOS
yum install gsm-toolsClick to copy -
Fedora
dnf install gsm-toolsClick to copy -
Raspbian
apt-get install libgsm-toolsClick to copy
libgsm-tools
User binaries for a GSM speech compressorThis package contains user binaries for libgsm, an implementation of the European GSM 06.10 provisional standard for full-rate speech transcoding, prI-ETS 300 036, which uses RPE/LTP (residual pulse excitation/long term prediction) coding at 13 kbit/s. GSM 06.10 compresses frames of 160 13-bit samples (8 kHz sampling rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility with typical UNIX applications, this implementation turns frames of 160 16-bit linear samples into 33-byte frames (1650 Bytes/s). The quality of the algorithm is good enough for reliable speaker recognition; even music often survives transcoding in recognizable form (given the bandwidth limitations of 8 kHz sampling rate). The interfaces offered are a front end modelled after compress(1), and a library API. Compression and decompression run faster than realtime on most SPARCstations. The implementation has been verified against the ETSI standard test patterns.
gsm-tools
GSM speech compressor toolsgsm
Contains runtime shared libraries for libgsm, an implementation ofthe European GSM 06.10 provisional standard for full-rate speech transcoding, prI-ETS 300 036, which uses RPE/LTP (residual pulse excitation/long term prediction) coding at 13 kbit/s. GSM 06.10 compresses frames of 162 13-bit samples (8 kHz sampling rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility with typical UNIX applications, our implementation turns frames of 160 16-bit linear samples into 33-byte frames (1650 Bytes/s). The quality of the algorithm is good enough for reliable speaker recognition; even music often survives transcoding in recognizable form (given the bandwidth limitations of 8 kHz sampling rate). The interfaces offered are a front end modelled after compress(1), and a library API. Compression and decompression run faster than realtime on most SPARCstations. The implementation has been verified against the ETSI standard test patterns.