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Formatting fixes 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2008-01-30 16:22:06 +00:00
parent 7bf09615a8
commit e7bcc4e96c
1 changed files with 9 additions and 9 deletions
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18
main/rtp.c
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@ -2610,10 +2610,10 @@ static int ast_rtp_senddigit_continuation(struct ast_rtp *rtp)
/* Setup packet to send */
rtpheader = (unsigned int *)data;
rtpheader[0] = htonl((2 << 30) | (1 << 23) | (rtp->send_payload << 16) | (rtp->seqno));
rtpheader[1] = htonl(rtp->lastdigitts);
rtpheader[2] = htonl(rtp->ssrc);
rtpheader[3] = htonl((rtp->send_digit << 24) | (0xa << 16) | (rtp->send_duration));
rtpheader[0] = htonl((2 << 30) | (1 << 23) | (rtp->send_payload << 16) | (rtp->seqno));
rtpheader[1] = htonl(rtp->lastdigitts);
rtpheader[2] = htonl(rtp->ssrc);
rtpheader[3] = htonl((rtp->send_digit << 24) | (0xa << 16) | (rtp->send_duration));
rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
/* Transmit */
@ -3059,7 +3059,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
res = sendto(rtp->s, (void *)rtpheader, f->datalen + hdrlen, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
if (res <0) {
if (res < 0) {
if (!rtp->nat || (rtp->nat && (ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
ast_debug(1, "RTP Transmission error of packet %d to %s:%d: %s\n", rtp->seqno, ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
} else if (((ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(rtp, FLAG_NAT_INACTIVE_NOWARN)) {
@ -3073,7 +3073,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
rtp->txoctetcount +=(res - hdrlen);
if (rtp->rtcp && rtp->rtcp->schedid < 1)
rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
}
if (rtp_debug_test_addr(&rtp->them))
@ -3179,7 +3179,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
while ((f = ast_smoother_read(rtp->smoother)) && (f->data))
ast_rtp_raw_write(rtp, f, codec);
} else {
/* Don't buffer outgoing frames; send them one-per-packet: */
/* Don't buffer outgoing frames; send them one-per-packet: */
if (_f->offset < hdrlen)
f = ast_frdup(_f); /*! \bug XXX this might never be free'd. Why do we do this? */
else
@ -3516,10 +3516,10 @@ static int p2p_callback_disable(struct ast_channel *chan, struct ast_rtp *rtp, i
static void p2p_set_bridge(struct ast_rtp *rtp0, struct ast_rtp *rtp1)
{
rtp_bridge_lock(rtp0);
rtp0->bridged = rtp1;
rtp0->bridged = rtp1;
rtp_bridge_unlock(rtp0);
return;
return;
}
/*! \brief Bridge loop for partial native bridge (packet2packet)