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Add remainder of rtp fixes, iax2 patch (bug #3961)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2005-05-14 23:57:44 +00:00
parent 060b71e52f
commit c936fc827c
4 changed files with 202 additions and 162 deletions
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121
rtp.c
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@@ -43,12 +43,6 @@
#define RTP_MTU 1200
#define TYPE_HIGH 0x0
#define TYPE_LOW 0x1
#define TYPE_SILENCE 0x2
#define TYPE_DONTSEND 0x3
#define TYPE_MASK 0x3
static int dtmftimeout = 3000; /* 3000 samples */
static int rtpstart = 0;
@@ -128,42 +122,6 @@ int ast_rtcp_fd(struct ast_rtp *rtp)
return -1;
}
static int g723_len(unsigned char buf)
{
switch(buf & TYPE_MASK) {
case TYPE_DONTSEND:
return 0;
break;
case TYPE_SILENCE:
return 4;
break;
case TYPE_HIGH:
return 24;
break;
case TYPE_LOW:
return 20;
break;
default:
ast_log(LOG_WARNING, "Badly encoded frame (%d)\n", buf & TYPE_MASK);
}
return -1;
}
static int g723_samples(unsigned char *buf, int maxlen)
{
int pos = 0;
int samples = 0;
int res;
while(pos < maxlen) {
res = g723_len(buf[pos]);
if (res <= 0)
break;
samples += 240;
pos += res;
}
return samples;
}
void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
{
rtp->data = data;
@@ -594,43 +552,9 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
switch(rtp->f.subclass) {
case AST_FORMAT_ULAW:
case AST_FORMAT_ALAW:
rtp->f.samples = rtp->f.datalen;
break;
case AST_FORMAT_SLINEAR:
rtp->f.samples = rtp->f.datalen / 2;
rtp->f.samples = ast_codec_get_samples(&rtp->f);
if (rtp->f.subclass == AST_FORMAT_SLINEAR)
ast_frame_byteswap_be(&rtp->f);
break;
case AST_FORMAT_GSM:
rtp->f.samples = 160 * (rtp->f.datalen / 33);
break;
case AST_FORMAT_ILBC:
rtp->f.samples = 240 * (rtp->f.datalen / 50);
break;
case AST_FORMAT_ADPCM:
case AST_FORMAT_G726:
rtp->f.samples = rtp->f.datalen * 2;
break;
case AST_FORMAT_G729A:
rtp->f.samples = rtp->f.datalen * 8;
break;
case AST_FORMAT_G723_1:
rtp->f.samples = g723_samples(rtp->f.data, rtp->f.datalen);
break;
case AST_FORMAT_SPEEX:
/* 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 *)(rtp->f.data))[7] & 0x1) * 8;
break;
default:
ast_log(LOG_NOTICE, "Unable to calculate samples for format %s\n", ast_getformatname(rtp->f.subclass));
break;
}
calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
} else {
/* Video -- samples is # of samples vs. 90000 */
@@ -1233,45 +1157,8 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
ms = calc_txstamp(rtp, &f->delivery);
/* Default prediction */
if (f->subclass < AST_FORMAT_MAX_AUDIO) {
pred = rtp->lastts + ms * 8;
switch(f->subclass) {
case AST_FORMAT_ULAW:
case AST_FORMAT_ALAW:
/* If we're within +/- 20ms from when where we
predict we should be, use that */
pred = rtp->lastts + f->datalen;
break;
case AST_FORMAT_ADPCM:
case AST_FORMAT_G726:
/* If we're within +/- 20ms from when where we
predict we should be, use that */
pred = rtp->lastts + f->datalen * 2;
break;
case AST_FORMAT_G729A:
pred = rtp->lastts + f->datalen * 8;
break;
case AST_FORMAT_GSM:
pred = rtp->lastts + (f->datalen * 160 / 33);
break;
case AST_FORMAT_ILBC:
pred = rtp->lastts + (f->datalen * 240 / 50);
break;
case AST_FORMAT_G723_1:
pred = rtp->lastts + g723_samples(f->data, f->datalen);
break;
case AST_FORMAT_SPEEX:
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));
}
pred = rtp->lastts + ast_codec_get_samples(f);
/* Re-calculate last TS */
rtp->lastts = rtp->lastts + ms * 8;
if (!f->delivery.tv_sec && !f->delivery.tv_usec) {