Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@120885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jeff Peeler
2008-06-06 16:39:20 +00:00
parent 6d12307629
commit a8281b2bcc
+11 -3
View File
@@ -1194,7 +1194,10 @@ static void temp_pvt_cleanup(void *);
/*! \brief A per-thread temporary pvt structure */
AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);
AST_THREADSTORAGE(ast_rtp_buf, ast_rtp_buf_init);
static void ts_ast_rtp_destroy(void *);
AST_THREADSTORAGE_CUSTOM(ts_audio_rtp, ts_audio_rtp_init, ts_ast_rtp_destroy);
AST_THREADSTORAGE_CUSTOM(ts_video_rtp, ts_video_rtp_init, ts_ast_rtp_destroy);
/*! \todo Move the sip_auth list to AST_LIST */
static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */
@@ -5063,7 +5066,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
#ifdef LOW_MEMORY
newaudiortp = ast_threadstorage_get(&ast_rtp_buf, ast_rtp_alloc_size());
newaudiortp = ast_threadstorage_get(&ts_audio_rtp, ast_rtp_alloc_size());
#else
newaudiortp = alloca(ast_rtp_alloc_size());
#endif
@@ -5072,7 +5075,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
ast_rtp_pt_clear(newaudiortp);
#ifdef LOW_MEMORY
newvideortp = ast_threadstorage_get(&ast_rtp_buf, ast_rtp_alloc_size());
newvideortp = ast_threadstorage_get(&ts_video_rtp, ast_rtp_alloc_size());
#else
newvideortp = alloca(ast_rtp_alloc_size());
#endif
@@ -5607,6 +5610,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
return 0;
}
static void ts_ast_rtp_destroy(void *data)
{
struct ast_rtp *tmp = data;
ast_rtp_destroy(tmp);
}
/*! \brief Add header to SIP message */
static int add_header(struct sip_request *req, const char *var, const char *value)