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authorJoshua Colp <jcolp@digium.com>2006-04-27 17:53:52 +0000
committerJoshua Colp <jcolp@digium.com>2006-04-27 17:53:52 +0000
commit6016d3703d9f697abedf58b4083d245eaf3def24 (patch)
treef291baae26e584037e7b1653766c8117e3ac98fc /rtp.c
parent3f0d9eeed024fdfdfe20d1150ed339b721a410aa (diff)
Change channel locking to API, nifty janitor project. (issue #7039 reported by Mithraen)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'rtp.c')
-rw-r--r--rtp.c52
1 files changed, 26 insertions, 26 deletions
diff --git a/rtp.c b/rtp.c
index a2646aa2f..e55c37901 100644
--- a/rtp.c
+++ b/rtp.c
@@ -739,11 +739,11 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
struct ast_rtp *vdestp, *vsrcp; /* Video RTP channels */
struct ast_rtp_protocol *destpr, *srcpr;
/* Lock channels */
- ast_mutex_lock(&dest->lock);
- while(ast_mutex_trylock(&src->lock)) {
- ast_mutex_unlock(&dest->lock);
+ ast_channel_lock(dest);
+ while(ast_channel_trylock(src)) {
+ ast_channel_unlock(dest);
usleep(1);
- ast_mutex_lock(&dest->lock);
+ ast_channel_lock(dest);
}
/* Find channel driver interfaces */
@@ -752,15 +752,15 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
if (!destpr) {
if (option_debug)
ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
- ast_mutex_unlock(&dest->lock);
- ast_mutex_unlock(&src->lock);
+ ast_channel_unlock(dest);
+ ast_channel_unlock(src);
return 0;
}
if (!srcpr) {
if (option_debug)
ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
- ast_mutex_unlock(&dest->lock);
- ast_mutex_unlock(&src->lock);
+ ast_channel_unlock(dest);
+ ast_channel_unlock(src);
return 0;
}
@@ -773,15 +773,15 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (!destp || !srcp) {
/* Somebody doesn't want to play... */
- ast_mutex_unlock(&dest->lock);
- ast_mutex_unlock(&src->lock);
+ ast_channel_unlock(dest);
+ ast_channel_unlock(src);
return 0;
}
ast_rtp_pt_copy(destp, srcp);
if (vdestp && vsrcp)
ast_rtp_pt_copy(vdestp, vsrcp);
- ast_mutex_unlock(&dest->lock);
- ast_mutex_unlock(&src->lock);
+ ast_channel_unlock(dest);
+ ast_channel_unlock(src);
if (option_debug)
ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
return 1;
@@ -1562,11 +1562,11 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
return AST_BRIDGE_FAILED_NOWARN;
/* Lock channels */
- ast_mutex_lock(&c0->lock);
- while(ast_mutex_trylock(&c1->lock)) {
- ast_mutex_unlock(&c0->lock);
+ ast_channel_lock(c0);
+ while(ast_channel_trylock(c1)) {
+ ast_channel_unlock(c0);
usleep(1);
- ast_mutex_lock(&c0->lock);
+ ast_channel_lock(c0);
}
/* Find channel driver interfaces */
@@ -1574,14 +1574,14 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
pr1 = get_proto(c1);
if (!pr0) {
ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
- ast_mutex_unlock(&c0->lock);
- ast_mutex_unlock(&c1->lock);
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
return AST_BRIDGE_FAILED;
}
if (!pr1) {
ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
- ast_mutex_unlock(&c0->lock);
- ast_mutex_unlock(&c1->lock);
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
return AST_BRIDGE_FAILED;
}
@@ -1598,8 +1598,8 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (!p0 || !p1) {
/* Somebody doesn't want to play... */
- ast_mutex_unlock(&c0->lock);
- ast_mutex_unlock(&c1->lock);
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
/* Get codecs from both sides */
@@ -1610,8 +1610,8 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
if (!(codec0 & codec1)) {
if (option_debug)
ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
- ast_mutex_unlock(&c0->lock);
- ast_mutex_unlock(&c1->lock);
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
}
@@ -1637,8 +1637,8 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
if (vp0)
ast_rtp_get_peer(vp0, &vac0);
}
- ast_mutex_unlock(&c0->lock);
- ast_mutex_unlock(&c1->lock);
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
/* External RTP Bridge up, now loop and see if something happes that force us to take the
media back to Asterisk */
cs[0] = c0;