mirror of
https://github.com/asterisk/asterisk.git
synced 2025-09-03 03:20:57 +00:00
d2776d4d45
This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable
errors caught by clang. Specifically:
* apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[],
qsmp_cmd_usage[]
* cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom"
* channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel"
* codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$"
* funcs/func_env.c:729: Fixed ast_str_append_substr.
* main/editline/np/strlcat.c: removed unused rcsid variable
* main/editline/np/strlcpy.c: removed unused rcsid variable
* main/security_events.c: removed unused TIMESTAMP_STR_LEN
* utils/conf2ael.c: removed unused cfextension_states
* utils/extconf.c: removed unused cfextension_states
Review: https://reviewboard.asterisk.org/r/4526
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4526.patch submitted by dkdegroot (License 6600)
........
Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 433694 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.