dect/asterisk
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- use '?' instead of if statements for assignment;

Browse source 
- fix indentation in a few places
- use a variable to store the 'other' channel, thus removing
  the need for some duplicated code;
- use '=' instead of memcpy to copy struct sockaddr_in



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@20197 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
rizzo 2006-04-14 23:30:18 +00:00
parent 3b962c0937
commit 937c5b6621
1 changed files with 16 additions and 37 deletions
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53
rtp.c
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@ -783,15 +783,9 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
/* Get audio and video interface (if native bridge is possible) */
destp = destpr->get_rtp_info(dest);
if (destpr->get_vrtp_info)
vdestp = destpr->get_vrtp_info(dest);
else
vdestp = NULL;
vdestp = (destpr->get_vrtp_info) ? destpr->get_vrtp_info(dest) : NULL;
srcp = srcpr->get_rtp_info(src);
if (srcpr->get_vrtp_info)
vsrcp = srcpr->get_vrtp_info(src);
else
vsrcp = NULL;
vsrcp = (srcpr->get_vrtp_info) ? srcpr->get_vrtp_info(src) : NULL;
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (!destp || !srcp) {
@ -832,13 +826,13 @@ void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
int i;
if (pt < 0 || pt > MAX_RTP_PT)
return; /* bogus payload type */
return; /* bogus payload type */
for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
strcasecmp(mimeType, mimeTypes[i].type) == 0) {
rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
return;
return;
}
}
}
@ -916,9 +910,8 @@ char* ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code)
int i;
for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
if (mimeTypes[i].payloadType.code == code && mimeTypes[i].payloadType.isAstFormat == isAstFormat) {
return mimeTypes[i].subtype;
}
if (mimeTypes[i].payloadType.code == code && mimeTypes[i].payloadType.isAstFormat == isAstFormat)
return mimeTypes[i].subtype;
}
return "";
}
@ -1119,7 +1112,7 @@ void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
{
memcpy(us, &rtp->us, sizeof(rtp->us));
*us = rtp->us;
}
void ast_rtp_stop(struct ast_rtp *rtp)
@ -1563,7 +1556,7 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
{
struct ast_frame *f;
struct ast_channel *who, *cs[3];
struct ast_channel *who, *other, *cs[3];
struct ast_rtp *p0, *p1; /* Audio RTP Channels */
struct ast_rtp *vp0, *vp1; /* Video RTP channels */
struct ast_rtp_protocol *pr0, *pr1;
@ -1615,15 +1608,9 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
/* Get audio and video interface (if native bridge is possible) */
p0 = pr0->get_rtp_info(c0);
if (pr0->get_vrtp_info)
vp0 = pr0->get_vrtp_info(c0);
else
vp0 = NULL;
vp0 = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0) : NULL;
p1 = pr1->get_rtp_info(c1);
if (pr1->get_vrtp_info)
vp1 = pr1->get_vrtp_info(c1);
else
vp1 = NULL;
vp1 = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1) : NULL;
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (!p0 || !p1) {
@ -1633,14 +1620,8 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
return AST_BRIDGE_FAILED_NOWARN;
}
/* Get codecs from both sides */
if (pr0->get_codec)
codec0 = pr0->get_codec(c0);
else
codec0 = 0;
if (pr1->get_codec)
codec1 = pr1->get_codec(c1);
else
codec1 = 0;
codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
if (pr0->get_codec && pr1->get_codec) {
/* Hey, we can't do reinvite if both parties speak different codecs */
if (!(codec0 & codec1)) {
@ -1751,9 +1732,11 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
continue;
}
f = ast_read(who);
other = (who == c0) ? c1 : c0; /* the other channel */
if (!f || ((f->frametype == AST_FRAME_DTMF) &&
(((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
/* breaking out of the bridge. */
*fo = f;
*rc = who;
if (option_debug)
@ -1770,7 +1753,7 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
} else if ((f->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
if ((f->subclass == AST_CONTROL_HOLD) || (f->subclass == AST_CONTROL_UNHOLD) ||
(f->subclass == AST_CONTROL_VIDUPDATE)) {
ast_indicate(who == c0 ? c1 : c0, f->subclass);
ast_indicate(other, f->subclass);
ast_frfree(f);
} else {
*fo = f;
@ -1783,11 +1766,7 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
(f->frametype == AST_FRAME_VOICE) ||
(f->frametype == AST_FRAME_VIDEO)) {
/* Forward voice or DTMF frames if they happen upon us */
if (who == c0) {
ast_write(c1, f);
} else if (who == c1) {
ast_write(c0, f);
}
ast_write(other, f);
}
ast_frfree(f);
}