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Version 0.3.0 from FTP

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2003-01-28 06:57:51 +00:00
parent 21203386be
commit 070c998a16
1 changed files with 5 additions and 5 deletions
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10
codecs/codec_speex.c
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@@ -10,7 +10,7 @@
* This program is free software, distributed under the terms of
* the GNU General Public License
*
* This work was motivated by Jeremy McNamera's
* This work was motivated by Jeremy McNamara
*/
#define TYPE_SILENCE 0x2
@@ -102,7 +102,7 @@ static struct ast_frame *lintospeex_sample()
f.subclass = AST_FORMAT_SLINEAR;
f.datalen = sizeof(slin_speex_ex);
/* Assume 8000 Hz */
f.timelen = sizeof(slin_speex_ex)/16;
f.samples = sizeof(slin_speex_ex)/2;
f.mallocd = 0;
f.offset = 0;
f.src = __PRETTY_FUNCTION__;
@@ -117,7 +117,7 @@ static struct ast_frame *speextolin_sample()
f.subclass = AST_FORMAT_SPEEX;
f.datalen = sizeof(speex_slin_ex);
/* All frames are 20 ms long */
f.timelen = 20;
f.samples = 160;
f.mallocd = 0;
f.offset = 0;
f.src = __PRETTY_FUNCTION__;
@@ -135,7 +135,7 @@ static struct ast_frame *speextolin_frameout(struct ast_translator_pvt *tmp)
tmp->f.subclass = AST_FORMAT_SLINEAR;
tmp->f.datalen = tmp->tail * 2;
/* Assume 8000 Hz */
tmp->f.timelen = tmp->tail / 8;
tmp->f.samples = tmp->tail;
tmp->f.mallocd = 0;
tmp->f.offset = AST_FRIENDLY_OFFSET;
tmp->f.src = __PRETTY_FUNCTION__;
@@ -220,7 +220,7 @@ static struct ast_frame *lintospeex_frameout(struct ast_translator_pvt *tmp)
speex_bits_pack(&tmp->bits, 15, 5);
len = speex_bits_write(&tmp->bits, (char *)tmp->outbuf, sizeof(tmp->outbuf));
tmp->f.datalen = len;
tmp->f.timelen = y * 20;
tmp->f.samples = y * 160;
#if 0
{
static int fd = -1;