From: Seungbae Shin <seungbae.shin@samsung.com>
Date: Fri, 26 Mar 2021 04:12:49 +0000 (+0900)
Subject: Fix SVACE defect
X-Git-Tag: submit/tizen/20210729.023123~109
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=f1a42837de9aca0f5ce906959624b104179b5077;p=platform%2Fcore%2Fapi%2Fwebrtc.git

Fix SVACE defect

- DEREF_OF_NULL.RET.ALLOC
- PASS_TO_PROC_AFTER_CHECK
- INVARIANT_RESULT.OP_ZERO
- NO_EFFECT
- UNREACHABLE_CODE

[Version] 0.1.132
[Issue Type] Svace

Change-Id: I48afb8b4a5842c3976d57fa190542a04e2b20935
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 17b3e952..ac3235ec 100644
--- a/packaging/capi-media-webrtc.spec
+++ b/packaging/capi-media-webrtc.spec
@@ -1,6 +1,6 @@
 Name:       capi-media-webrtc
 Summary:    A WebRTC library in Tizen Native API
-Version:    0.1.131
+Version:    0.1.132
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_sink.c b/src/webrtc_sink.c
index dbaee98c..10ac5bd6 100644
--- a/src/webrtc_sink.c
+++ b/src/webrtc_sink.c
@@ -442,7 +442,7 @@ int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad)
 	gchar *track_name;
 	webrtc_gst_slot_s *sink;
 	GstElement *decodebin;
-	GstPad *sink_pad;
+	GstPad *sink_pad = NULL;
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(src_pad == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "src_pad is NULL");
@@ -500,7 +500,8 @@ int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad)
 	return WEBRTC_ERROR_NONE;
 
 error_before_insert:
-	gst_object_unref(sink_pad);
+	if (sink_pad)
+		gst_object_unref(sink_pad);
 	g_free(track_name);
 	g_free(sink);
 
diff --git a/test/webrtc_test.c b/test/webrtc_test.c
index f2929565..80d92655 100644
--- a/test/webrtc_test.c
+++ b/test/webrtc_test.c
@@ -334,7 +334,7 @@ static void __release_packet_source(int conn_idx, unsigned int source_idx)
 {
 	if (conn_idx < 0 || conn_idx >= MAX_CONNECTION_LEN)
 		return;
-	if (source_idx < 0 || source_idx >= MAX_MEDIA_PACKET_SOURCE_LEN)
+	if (source_idx >= MAX_MEDIA_PACKET_SOURCE_LEN)
 		return;
 
 	media_packet_source_s *packet_source = &g_conns[conn_idx].packet_sources[source_idx];
@@ -796,7 +796,7 @@ static void _webrtc_set_stun_server(int index, char *uri)
 	if (!uri)
 		return;
 
-	webrtc_set_stun_server(g_conns[index].webrtc, uri);
+	ret = webrtc_set_stun_server(g_conns[index].webrtc, uri);
 	if (ret != WEBRTC_ERROR_NONE)
 		g_print("failed to webrtc_set_stun_server(), uri[%s], ret[0x%x]\n", uri, ret);
 	else
@@ -853,7 +853,8 @@ static void __file_dump(const gchar *file_name, void *src_buffer, int size)
 
 end:
 	g_free(full_path);
-	close(fd);
+	if (fd != -1)
+		close(fd);
 }
 
 static void __data_channel_message_cb(webrtc_data_channel_h channel, webrtc_data_channel_type_e type, void *message, void *user_data)