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First pass at LPC10

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2004-09-15 18:40:07 +00:00
parent 9a387e5a7a
commit ce68497243
1 changed files with 11 additions and 1 deletions
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12
rtp.c
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@@ -549,8 +549,12 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
rtp->f.samples = g723_samples(rtp->f.data, rtp->f.datalen);
break;
case AST_FORMAT_SPEEX:
rtp->f.samples = 160;
/* assumes that the RTP packet contained one Speex frame */
rtp->f.samples = 160;
break;
case AST_FORMAT_LPC10:
rtp->f.samples = 22 * 8;
rtp->f.samples += (((char *)(f->data))[7] & 0x1) * 8;
break;
default:
ast_log(LOG_NOTICE, "Unable to calculate samples for format %s\n", ast_getformatname(rtp->f.subclass));
@@ -1082,6 +1086,11 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
pred = rtp->lastts + 160;
/* assumes that the RTP packet contains one Speex frame */
break;
case AST_FORMAT_LPC10:
/* assumes that the RTP packet contains one LPC10 frame */
pred = rtp->lastts + 22 * 8;
pred += (((char *)(f->data))[7] & 0x1) * 8;
break;
default:
ast_log(LOG_WARNING, "Not sure about timestamp format for codec format %s\n", ast_getformatname(f->subclass));
}
@@ -1245,6 +1254,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
case AST_FORMAT_H261:
case AST_FORMAT_H263:
case AST_FORMAT_G723_1:
case AST_FORMAT_LPC10:
case AST_FORMAT_SPEEX:
/* Don't buffer outgoing frames; send them one-per-packet: */
if (_f->offset < hdrlen) {