[asterisk-commits] twilson: trunk r221278 - in /trunk: channels/ include/asterisk/ main/ res/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 30 13:21:08 CDT 2009
Author: twilson
Date: Wed Sep 30 13:21:03 2009
New Revision: 221278
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=221278
Log:
Use rtp properties instead of adding a callback
Thanks, Josh.
Modified:
trunk/channels/chan_sip.c
trunk/include/asterisk/rtp_engine.h
trunk/main/rtp_engine.c
trunk/res/res_rtp_asterisk.c
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=221278&r1=221277&r2=221278
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Sep 30 13:21:03 2009
@@ -5191,11 +5191,9 @@
if (dialog->rtp) { /* Audio */
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
+ ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC));
ast_rtp_instance_set_timeout(dialog->rtp, peer->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
- if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
- ast_rtp_instance_set_constantssrc(dialog->rtp);
- }
/* Set Frame packetization */
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
@@ -5203,9 +5201,7 @@
if (dialog->vrtp) { /* Video */
ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
- if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
- ast_rtp_instance_set_constantssrc(dialog->vrtp);
- }
+ ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC));
}
if (dialog->trtp) { /* Realtime text */
ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
@@ -20495,13 +20491,11 @@
ast_debug(1, "No compatible codecs for this SIP call.\n");
return -1;
}
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
- if (p->rtp) {
- ast_rtp_instance_set_constantssrc(p->rtp);
- }
- if (p->vrtp) {
- ast_rtp_instance_set_constantssrc(p->vrtp);
- }
+ if (p->rtp) {
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC));
+ }
+ if (p->vrtp) {
+ ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC));
}
} else { /* No SDP in invite, call control session */
p->jointcapability = p->capability;
Modified: trunk/include/asterisk/rtp_engine.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/rtp_engine.h?view=diff&rev=221278&r1=221277&r2=221278
==============================================================================
--- trunk/include/asterisk/rtp_engine.h (original)
+++ trunk/include/asterisk/rtp_engine.h Wed Sep 30 13:21:03 2009
@@ -94,6 +94,8 @@
AST_RTP_PROPERTY_RTCP,
/*! Maximum number of RTP properties supported */
AST_RTP_PROPERTY_MAX,
+ /*! Don't force a new SSRC on new source */
+ AST_RTP_PROPERTY_CONSTANT_SSRC,
};
/*! Additional RTP options */
@@ -1185,23 +1187,6 @@
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance);
/*!
- * \brief Mark an RTP instance not to update SSRC on a new source
- *
- * \param instance Instance to update
- *
- * Example usage:
- *
- * \code
- * ast_rtp_instance_set_constantssrc(instance);
- * \endcode
- *
- * This sets the indicated instance to not update the RTP SSRC when new_source
- * is called.
- *
- * \since 1.6.3
- */
-void ast_rtp_instance_set_constantssrc(struct ast_rtp_instance *instance);
-/*!
* \brief Indicate a new source of audio has dropped in
*
* \param instance Instance that the new media source is feeding into
Modified: trunk/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/rtp_engine.c?view=diff&rev=221278&r1=221277&r2=221278
==============================================================================
--- trunk/main/rtp_engine.c (original)
+++ trunk/main/rtp_engine.c Wed Sep 30 13:21:03 2009
@@ -724,13 +724,6 @@
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance)
{
return instance->dtmf_mode;
-}
-
-void ast_rtp_instance_set_constantssrc(struct ast_rtp_instance *instance)
-{
- if (instance->engine->constant_ssrc_set) {
- instance->engine->constant_ssrc_set(instance);
- }
}
void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
Modified: trunk/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_rtp_asterisk.c?view=diff&rev=221278&r1=221277&r2=221278
==============================================================================
--- trunk/res/res_rtp_asterisk.c (original)
+++ trunk/res/res_rtp_asterisk.c Wed Sep 30 13:21:03 2009
@@ -103,7 +103,6 @@
#define FLAG_NAT_INACTIVE_NOWARN (1 << 1)
#define FLAG_NEED_MARKER_BIT (1 << 3)
#define FLAG_DTMF_COMPENSATE (1 << 4)
-#define FLAG_CONSTANT_SSRC (1 << 5)
/*! \brief RTP session description */
struct ast_rtp {
@@ -254,7 +253,6 @@
static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit);
static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit);
static void ast_rtp_new_source(struct ast_rtp_instance *instance);
-static void ast_rtp_set_constantssrc(struct ast_rtp_instance *instance);
static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
@@ -277,7 +275,6 @@
.dtmf_begin = ast_rtp_dtmf_begin,
.dtmf_end = ast_rtp_dtmf_end,
.new_source = ast_rtp_new_source,
- .constant_ssrc_set = ast_rtp_set_constantssrc,
.write = ast_rtp_write,
.read = ast_rtp_read,
.prop_set = ast_rtp_prop_set,
@@ -656,13 +653,6 @@
return 0;
}
-void ast_rtp_set_constantssrc(struct ast_rtp_instance *instance)
-{
- struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
-
- ast_set_flag(rtp, FLAG_CONSTANT_SSRC);
-}
-
static void ast_rtp_new_source(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
@@ -670,7 +660,8 @@
/* We simply set this bit so that the next packet sent will have the marker bit turned on */
ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
- if (!ast_test_flag(rtp, FLAG_CONSTANT_SSRC)) {
+ if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_CONSTANT_SSRC)) {
+ ast_log(LOG_ERROR, "Changing ssrc\n");
rtp->ssrc = ast_random();
}
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