gsm_ilbc_lpc10_codecs
所属分类：IP电话/视频会议
开发工具：Unix_Linux
文件大小：416KB
下载次数：47
上传日期：2008-06-30 13:21:29
上 传 者：
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说明： Voip 语音压缩解压算法 gsm iLbc lpc10
(Voip voice compression decompression algorithm gsm iLbc lpc10)
文件列表:
codecs\adpcm_slin_ex.h (775, 2006-02-15)
codecs\codec_adpcm.c (9071, 2006-08-21)
codecs\codec_alaw.c (4342, 2006-08-21)
codecs\codec_a_mu.c (3816, 2006-08-21)
codecs\codec_g726.c (25221, 2006-08-21)
codecs\codec_gsm.c (7349, 2007-01-31)
codecs\codec_ilbc.c (6310, 2007-05-24)
codecs\codec_lpc10.c (8396, 2006-09-09)
codecs\codec_speex.c (15716, 2007-05-24)
codecs\codec_ulaw.c (4454, 2006-08-21)
codecs\codec_zap.c (11310, 2007-08-31)
codecs\g726_slin_ex.h (774, 2006-02-15)
codecs\gsm_slin_ex.h (429, 2006-02-15)
codecs\ilbc_slin_ex.h (523, 2006-02-15)
codecs\log2comp.h (1461, 2006-02-15)
codecs\lpc10_slin_ex.h (273, 2006-02-15)
codecs\Makefile (1273, 2007-01-05)
codecs\slin_adpcm_ex.h (941, 2006-02-15)
codecs\slin_g726_ex.h (940, 2006-02-15)
codecs\slin_gsm_ex.h (1534, 2006-02-15)
codecs\slin_ilbc_ex.h (1534, 2006-02-15)
codecs\slin_lpc10_ex.h (970, 2006-02-15)
codecs\slin_speex_ex.h (32286, 2006-02-15)
codecs\slin_ulaw_ex.h (939, 2006-02-15)
codecs\speex_slin_ex.h (469, 2006-02-15)
codecs\ulaw_slin_ex.h (772, 2006-02-15)
codecs\lpc10\analys.c (24051, 2006-10-14)
codecs\lpc10\bsynz.c (14866, 2006-10-14)
codecs\lpc10\chanwr.c (7214, 2006-10-14)
codecs\lpc10\dcbias.c (2660, 2006-10-14)
codecs\lpc10\decode.c (20600, 2006-10-14)
codecs\lpc10\deemp.c (4205, 2006-10-14)
codecs\lpc10\difmag.c (3668, 2006-10-14)
codecs\lpc10\dyptrk.c (13350, 2006-10-14)
codecs\lpc10\encode.c (12330, 2006-10-14)
codecs\lpc10\energy.c (2550, 2006-10-14)
codecs\lpc10\f2c.h (10072, 2006-02-15)
codecs\lpc10\f2clib.c (1186, 2006-02-15)
codecs\lpc10\ham84.c (3547, 2006-10-14)
codecs\lpc10\hp100.c (4689, 2006-10-14)
... ...
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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