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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
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+11
-3
@@ -1194,7 +1194,10 @@ static void temp_pvt_cleanup(void *);
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/*! \brief A per-thread temporary pvt structure */
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AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);
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AST_THREADSTORAGE(ast_rtp_buf, ast_rtp_buf_init);
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static void ts_ast_rtp_destroy(void *);
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AST_THREADSTORAGE_CUSTOM(ts_audio_rtp, ts_audio_rtp_init, ts_ast_rtp_destroy);
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AST_THREADSTORAGE_CUSTOM(ts_video_rtp, ts_video_rtp_init, ts_ast_rtp_destroy);
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/*! \todo Move the sip_auth list to AST_LIST */
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static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */
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@@ -5063,7 +5066,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
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#ifdef LOW_MEMORY
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newaudiortp = ast_threadstorage_get(&ast_rtp_buf, ast_rtp_alloc_size());
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newaudiortp = ast_threadstorage_get(&ts_audio_rtp, ast_rtp_alloc_size());
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#else
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newaudiortp = alloca(ast_rtp_alloc_size());
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#endif
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@@ -5072,7 +5075,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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ast_rtp_pt_clear(newaudiortp);
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#ifdef LOW_MEMORY
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newvideortp = ast_threadstorage_get(&ast_rtp_buf, ast_rtp_alloc_size());
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newvideortp = ast_threadstorage_get(&ts_video_rtp, ast_rtp_alloc_size());
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#else
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newvideortp = alloca(ast_rtp_alloc_size());
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#endif
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@@ -5607,6 +5610,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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return 0;
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}
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static void ts_ast_rtp_destroy(void *data)
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{
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struct ast_rtp *tmp = data;
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ast_rtp_destroy(tmp);
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}
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/*! \brief Add header to SIP message */
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static int add_header(struct sip_request *req, const char *var, const char *value)
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