文件列表:

add-test (0, 2000-01-24)
add-test\add_test.c (7363, 1994-05-11)
add-test\add_test.dta (12203, 1994-05-11)
ChangeLog.TXT (2167, 1994-12-31)
COPYRIGHT.TXT (706, 1994-11-30)
inc (0, 2000-01-24)
inc\config.h (1282, 1994-05-11)
inc\gsm.h (1675, 1994-12-31)
inc\private.h (7722, 1994-11-30)
inc\proto.h (1671, 1992-10-28)
inc\toast.h (2199, 1994-11-30)
inc\unproto.h (494, 1992-10-28)
Install.TXT (2639, 1994-11-30)
Machines.TXT (451, 1994-05-11)
Makefile (11001, 1994-05-11)
man (0, 2000-01-24)
Manifest.TXT (1248, 1992-10-29)
man\bitter.1 (1835, 1994-11-30)
man\gsm.3 (2922, 1994-12-31)
man\gsm_explode.3 (1286, 1994-12-31)
man\gsm_option.3 (2455, 1994-12-31)
man\gsm_print.3 (1599, 1994-12-31)
man\toast.1 (3666, 1994-11-30)
src (0, 2000-01-24)
src\add.c (5784, 1994-12-31)
src\code.c (2653, 1994-05-11)
src\debug.c (1619, 1994-05-11)
src\decode.c (1632, 1992-10-28)
src\gsm_create.c (1019, 1994-05-11)
src\gsm_decode.c (3684, 1992-10-28)
src\gsm_destroy.c (594, 1994-11-30)
src\gsm_encode.c (5053, 1992-10-28)
src\gsm_explode.c (4443, 1992-10-28)
src\gsm_implode.c (6042, 1992-10-28)
src\gsm_option.c (787, 1992-10-28)
src\gsm_print.c (5097, 1992-10-28)
src\long_term.c (15847, 1994-05-11)
src\lpc.c (7429, 1994-12-31)
... ...

/* * 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. */ 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

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