Merge topic 'misc_rtp_tweaks' into 13

* changes:
  rtp_engine.c: Must protect mime_types_len with mime_types_lock.
  res_pjsip_sdp_rtp.c: Fixup some whitespace.

main/rtp_engine.c

@@ -603,6 +603,7 @@ void ast_rtp_codecs_payloads_destroy(struct ast_rtp_codecs *codecs)
 void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance)
 {
 	ast_rtp_codecs_payloads_destroy(codecs);
+	ast_rtp_codecs_payloads_initialize(codecs);
 
 	if (instance && instance->engine && instance->engine->payload_set) {
 		int i;
@@ -610,8 +611,6 @@ void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp
 			instance->engine->payload_set(instance, i, 0, NULL, 0);
 		}
 	}
-
-	ast_rtp_codecs_payloads_initialize(codecs);
 }
 
 void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance)
@@ -1716,12 +1715,16 @@ static void rtp_engine_mime_type_cleanup(int i)
 
 static void set_next_mime_type(struct ast_format *format, int rtp_code, const char *type, const char *subtype, unsigned int sample_rate)
 {
-	int x = mime_types_len;
-	if (ARRAY_LEN(ast_rtp_mime_types) == mime_types_len) {
+	int x;
+
+	ast_rwlock_wrlock(&mime_types_lock);
+
+	x = mime_types_len;
+	if (ARRAY_LEN(ast_rtp_mime_types) <= x) {
+		ast_rwlock_unlock(&mime_types_lock);
 		return;
 	}
 
-	ast_rwlock_wrlock(&mime_types_lock);
 	/* Make sure any previous value in ast_rtp_mime_types is cleaned up */
 	memset(&ast_rtp_mime_types[x], 0, sizeof(struct ast_rtp_mime_type));	
 	if (format) {
@@ -1734,6 +1737,7 @@ static void set_next_mime_type(struct ast_format *format, int rtp_code, const ch
 	ast_copy_string(ast_rtp_mime_types[x].subtype, subtype, sizeof(ast_rtp_mime_types[x].subtype));
 	ast_rtp_mime_types[x].sample_rate = sample_rate;
 	mime_types_len++;
+
 	ast_rwlock_unlock(&mime_types_lock);
 }
 
@@ -2095,6 +2099,7 @@ static void rtp_engine_shutdown(void)
 			rtp_engine_mime_type_cleanup(x);
 		}
 	}
+	mime_types_len = 0;
 	ast_rwlock_unlock(&mime_types_lock);
 }
 

res/res_pjsip_sdp_rtp.c

@@ -237,13 +237,13 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
 		}
 
 		ast_copy_pj_str(name, &rtpmap->enc_name, sizeof(name));
-                if (strcmp(name,"telephone-event") == 0) {
-                        tel_event++;
-                }
+		if (strcmp(name, "telephone-event") == 0) {
+			tel_event++;
+		}
 
 		ast_copy_pj_str(media, (pj_str_t*)&stream->desc.media, sizeof(media));
-		ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, NULL, pj_strtoul(&stream->desc.fmt[i]),
-							     media, name, options, rtpmap->clock_rate);
+		ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, NULL,
+			pj_strtoul(&stream->desc.fmt[i]), media, name, options, rtpmap->clock_rate);
 		/* Look for an optional associated fmtp attribute */
 		if (!(attr = pjmedia_sdp_media_find_attr2(stream, "fmtp", &rtpmap->pt))) {
 			continue;
@@ -270,8 +270,8 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
 			}
 		}
 	}
-	if ((tel_event==0) && (session->endpoint->dtmf == AST_SIP_DTMF_AUTO)) {
-                ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
+	if (!tel_event && (session->endpoint->dtmf == AST_SIP_DTMF_AUTO)) {
+		ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
 	}
 	/* Get the packetization, if it exists */
 	if ((attr = pjmedia_sdp_media_find_attr2(stream, "ptime", NULL))) {
@@ -329,7 +329,7 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
 	}
 
 	ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
-				     session_media->rtp);
+		session_media->rtp);
 
 	ast_format_cap_append_from_cap(session->req_caps, joint, AST_MEDIA_TYPE_UNKNOWN);
 
@@ -1137,8 +1137,12 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 	/* Add non-codec formats */
 	if (media_type != AST_MEDIA_TYPE_VIDEO) {
 		for (index = 1LL; index <= AST_RTP_MAX; index <<= 1) {
-			if (!(noncodec & index) || (rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp),
-											   0, NULL, index)) == -1) {
+			if (!(noncodec & index)) {
+				continue;
+			}
+			rtp_code = ast_rtp_codecs_payload_code(
+				ast_rtp_instance_get_codecs(session_media->rtp), 0, NULL, index);
+			if (rtp_code == -1) {
 				continue;
 			}