Reduce memory consumption significantly for users of the RTP engine API by storing only the payloads present and in use instead of every possible one.

Review: https://reviewboard.asterisk.org/r/2052/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 21 insertions, 27 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b002254ef..b65390aac 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -9450,7 +9450,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 
 	int peernoncodeccapability = 0, vpeernoncodeccapability = 0, tpeernoncodeccapability = 0;
 
-	struct ast_rtp_codecs *newaudiortp = NULL, *newvideortp = NULL, *newtextrtp = NULL;
+	struct ast_rtp_codecs newaudiortp = { 0, }, newvideortp = { 0, }, newtextrtp = { 0, };
 	struct ast_format_cap *newjointcapability = ast_format_cap_alloc_nolock(); /* Negotiated capability */
 	struct ast_format_cap *newpeercapability = ast_format_cap_alloc_nolock();
 	int newnoncodeccapability;
@@ -9487,8 +9487,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		goto process_sdp_cleanup;
 	}
 
-	if (!(newaudiortp = ast_calloc(1, sizeof(*newaudiortp))) || !(newvideortp = ast_calloc(1, sizeof(*newvideortp))) ||
-	    !(newtextrtp = ast_calloc(1, sizeof(*newtextrtp)))) {
+	if (ast_rtp_codecs_payloads_initialize(&newaudiortp) || ast_rtp_codecs_payloads_initialize(&newvideortp) ||
+	    ast_rtp_codecs_payloads_initialize(&newtextrtp)) {
 		res = -1;
 		goto process_sdp_cleanup;
 	}
@@ -9532,11 +9532,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 			if (process_sdp_a_sendonly(value, &sendonly)) {
 				processed = TRUE;
 			}
-			else if (process_sdp_a_audio(value, p, newaudiortp, &last_rtpmap_codec))
+			else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec))
 				processed = TRUE;
-			else if (process_sdp_a_video(value, p, newvideortp, &last_rtpmap_codec))
+			else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec))
 				processed = TRUE;
-			else if (process_sdp_a_text(value, p, newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
+			else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
 				processed = TRUE;
 			else if (process_sdp_a_image(value, p))
 				processed = TRUE;
@@ -9650,7 +9650,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 						ast_verbose("Found RTP audio format %d\n", codec);
 					}
 
-					ast_rtp_codecs_payloads_set_m_type(newaudiortp, NULL, codec);
+					ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec);
 				}
 			} else {
 				ast_log(LOG_WARNING, "Rejecting audio media offer due to invalid or unsupported syntax: %s\n", m);
@@ -9722,7 +9722,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 					if (debug) {
 						ast_verbose("Found RTP video format %d\n", codec);
 					}
-					ast_rtp_codecs_payloads_set_m_type(newvideortp, NULL, codec);
+					ast_rtp_codecs_payloads_set_m_type(&newvideortp, NULL, codec);
 				}
 			} else {
 				ast_log(LOG_WARNING, "Rejecting video media offer due to invalid or unsupported syntax: %s\n", m);
@@ -9786,7 +9786,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 					if (debug) {
 						ast_verbose("Found RTP text format %d\n", codec);
 					}
-					ast_rtp_codecs_payloads_set_m_type(newtextrtp, NULL, codec);
+					ast_rtp_codecs_payloads_set_m_type(&newtextrtp, NULL, codec);
 				}
 			} else {
 				ast_log(LOG_WARNING, "Rejecting text stream offer due to invalid or unsupported syntax: %s\n", m);
@@ -9904,7 +9904,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 					} else if (!processed_crypto && process_crypto(p, p->rtp, &p->srtp, value)) {
 						processed_crypto = TRUE;
 						processed = TRUE;
-					} else if (process_sdp_a_audio(value, p, newaudiortp, &last_rtpmap_codec)) {
+					} else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) {
 						processed = TRUE;
 					}
 				}
@@ -9915,7 +9915,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 					} else if (!processed_crypto && process_crypto(p, p->vrtp, &p->vsrtp, value)) {
 						processed_crypto = TRUE;
 						processed = TRUE;
-					} else if (process_sdp_a_video(value, p, newvideortp, &last_rtpmap_codec)) {
+					} else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) {
 						processed = TRUE;
 					}
 				}
@@ -9923,7 +9923,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 				else if (text) {
 					if (process_sdp_a_ice(value, p, p->trtp)) {
 						processed = TRUE;
-					} if (process_sdp_a_text(value, p, newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
+					} if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
 						processed = TRUE;
 					} else if (!processed_crypto && process_crypto(p, p->trtp, &p->tsrtp, value)) {
 						processed_crypto = TRUE;
@@ -9996,9 +9996,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 	}
 
 	/* Now gather all of the codecs that we are asked for: */
-	ast_rtp_codecs_payload_formats(newaudiortp, peercapability, &peernoncodeccapability);
-	ast_rtp_codecs_payload_formats(newvideortp, vpeercapability, &vpeernoncodeccapability);
-	ast_rtp_codecs_payload_formats(newtextrtp, tpeercapability, &tpeernoncodeccapability);
+	ast_rtp_codecs_payload_formats(&newaudiortp, peercapability, &peernoncodeccapability);
+	ast_rtp_codecs_payload_formats(&newvideortp, vpeercapability, &vpeernoncodeccapability);
+	ast_rtp_codecs_payload_formats(&newtextrtp, tpeercapability, &tpeernoncodeccapability);
 
 	ast_format_cap_append(newpeercapability, peercapability);
 	ast_format_cap_append(newpeercapability, vpeercapability);
@@ -10061,7 +10061,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 					    ast_sockaddr_stringify(sa));
 			}
 
-			ast_rtp_codecs_payloads_copy(newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
+			ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
 			/* Ensure RTCP is enabled since it may be inactive
 			   if we're coming back from a T.38 session */
 			ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
@@ -10108,7 +10108,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 				ast_verbose("Peer video RTP is at port %s\n",
 					    ast_sockaddr_stringify(vsa));
 			}
-			ast_rtp_codecs_payloads_copy(newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
+			ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
 		} else {
 			ast_rtp_instance_stop(p->vrtp);
 			if (debug)
@@ -10132,7 +10132,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 			} else {
 				p->red = 0;
 			}
-			ast_rtp_codecs_payloads_copy(newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
+			ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
 		} else {
 			ast_rtp_instance_stop(p->trtp);
 			if (debug)
@@ -10250,15 +10250,9 @@ process_sdp_cleanup:
 	if (res) {
 		offered_media_list_destroy(p);
 	}
-	if (newtextrtp) {
-		ast_free(newtextrtp);
-	}
-	if (newvideortp) {
-		ast_free(newvideortp);
-	}
-	if (newaudiortp) {
-		ast_free(newaudiortp);
-	}
+	ast_rtp_codecs_payloads_destroy(&newtextrtp);
+	ast_rtp_codecs_payloads_destroy(&newvideortp);
+	ast_rtp_codecs_payloads_destroy(&newaudiortp);
 	ast_format_cap_destroy(peercapability);
 	ast_format_cap_destroy(vpeercapability);
 	ast_format_cap_destroy(tpeercapability);