Files could not be expanded: network:telephony:asterisk/libgsm: project 'network:telephony:asterisk' does not exist

GSM 06.10 Lossy Speech Compressor Library and Utilities

Edit Package libgsm

This package is based on the package 'libgsm' from project 'network:telephony'.

This package is based on the package 'libgsm' from project 'openSUSE:Factory'.

Contains libraries and binaries for a GSM speech compressor. libgsm
contains a standard 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,
which is 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 modeled after compress(1) and a
library API. Compression and decompression run faster than real-time
on most SPARC stations. The implementation has been verified against
the ETSI standard test patterns.

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