文件列表:

gsmlib\ChangeLog (3206, 2001-11-20)
gsmlib\code.c (2547, 2001-11-20)
gsmlib\config.h (1393, 2001-11-20)
gsmlib\COPYRIGHT (690, 2001-11-20)
gsmlib\debug.c (1543, 2001-11-20)
gsmlib\decode.c (1569, 2001-11-20)
gsmlib\gsm.h (2145, 2001-11-20)
gsmlib\gsmadd.c (5554, 2005-03-10)
gsmlib\gsm_create.c (855, 2001-11-20)
gsmlib\gsm_decode.c (10665, 2001-11-20)
gsmlib\gsm_destroy.c (568, 2001-11-20)
gsmlib\gsm_encode.c (11425, 2001-11-20)
gsmlib\gsm_explode.c (11163, 2001-11-20)
gsmlib\gsm_implode.c (12994, 2001-11-20)
gsmlib\gsm_option.c (1204, 2001-11-20)
gsmlib\gsm_print.c (4930, 2001-11-20)
gsmlib\install (3143, 2001-11-20)
gsmlib\long_term.c (23586, 2001-11-20)
gsmlib\lpc.c (7088, 2001-11-20)
gsmlib\machines (440, 2001-11-20)
gsmlib\Makefile.am (1332, 2005-03-10)
gsmlib\Makefile.in (20213, 2005-03-23)
gsmlib\manifest (1277, 2001-11-20)
gsmlib\preprocess.c (2512, 2001-11-20)
gsmlib\private.h (7782, 2001-11-20)
gsmlib\proto.h (1606, 2001-11-20)
gsmlib\rpe.c (11059, 2001-11-20)
gsmlib\short_term.c (10401, 2001-11-20)
gsmlib\table.c (2146, 2001-11-20)
gsmlib\toast.h (2179, 2001-11-20)
gsmlib\unproto.h (471, 2001-11-20)
gsmlib (0, 2005-06-03)

GSM 06.10 13 kbit/s RPE/LTP speech compression available -------------------------------------------------------- The Communications and Operating Systems Research Group (KBS) at the Technische Universitaet Berlin is currently working on a set of UNIX-based tools for computer-mediated telecooperation that will be made freely available. As part of this effort we are publishing 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, 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. Jutta Degener (jutta@cs.tu-berlin.de) Carsten Bormann (cabo@cs.tu-berlin.de) Communications and Operating Systems Research Group, TU Berlin Fax: +49.30.31425156, Phone: +49.30.31424315 -- Copyright 1992 by Jutta Degener and Carsten Bormann, Technische Universitaet Berlin. See the accompanying file "COPYRIGHT" for details. THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.

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