From: Sangchul Lee <sc11.lee@samsung.com>
Date: Wed, 3 Jan 2024 03:02:18 +0000 (+0900)
Subject: Add locking mutex for state
X-Git-Tag: accepted/tizen/unified/20240105.013006^0
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=a4d1ffdec1da9c091d5ef43996318896e7ea967d;p=platform%2Fcore%2Fapi%2Fwebrtc.git

Add locking mutex for state

[Version] 0.4.29
[Issue Type] Coverity defect (MISSING_LOCK)

Change-Id: If29613750c55c1179a3393de65f52866be4997b8
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 6ff3c348..7aada97d 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:    0.4.28
+Version:    0.4.29
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc.c b/src/webrtc.c
index f5927e5d..221d06fd 100644
--- a/src/webrtc.c
+++ b/src/webrtc.c
@@ -748,11 +748,12 @@ int webrtc_screen_source_set_crop(webrtc_h webrtc, unsigned int source_id, int x
 	RET_VAL_IF(y < 0, WEBRTC_ERROR_INVALID_PARAMETER, "y < 0");
 	RET_VAL_IF(width == 0, WEBRTC_ERROR_INVALID_PARAMETER, "width is 0");
 	RET_VAL_IF(height == 0, WEBRTC_ERROR_INVALID_PARAMETER, "height is 0");
-	RET_VAL_IF(_webrtc->state == WEBRTC_STATE_IDLE, WEBRTC_ERROR_INVALID_STATE, "the state should not be IDLE");
-	RET_ERR_IF_PRIVILEGE_IS_NOT_ALLOWED(_WEBRTC_PRIVILEGE_PLATFORM);
 
 	locker = g_mutex_locker_new(&_webrtc->mutex);
 
+	RET_VAL_IF(_webrtc->state == WEBRTC_STATE_IDLE, WEBRTC_ERROR_INVALID_STATE, "the state should not be IDLE");
+	RET_ERR_IF_PRIVILEGE_IS_NOT_ALLOWED(_WEBRTC_PRIVILEGE_PLATFORM);
+
 #ifdef TIZEN_FEATURE_UI
 	return _set_screen_source_crop(_webrtc, source_id, x, y, width, height);
 #else
@@ -768,11 +769,12 @@ int webrtc_screen_source_unset_crop(webrtc_h webrtc, unsigned int source_id)
 
 	RET_VAL_IF(_webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
-	RET_VAL_IF(_webrtc->state == WEBRTC_STATE_IDLE, WEBRTC_ERROR_INVALID_STATE, "the state should not be IDLE");
-	RET_ERR_IF_PRIVILEGE_IS_NOT_ALLOWED(_WEBRTC_PRIVILEGE_PLATFORM);
 
 	locker = g_mutex_locker_new(&_webrtc->mutex);
 
+	RET_VAL_IF(_webrtc->state == WEBRTC_STATE_IDLE, WEBRTC_ERROR_INVALID_STATE, "the state should not be IDLE");
+	RET_ERR_IF_PRIVILEGE_IS_NOT_ALLOWED(_WEBRTC_PRIVILEGE_PLATFORM);
+
 #ifdef TIZEN_FEATURE_UI
 	return _unset_screen_source_crop(_webrtc, source_id);
 #else
diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index 0cd747f9..c35e1cc0 100644
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -1452,12 +1452,15 @@ static void __webrtcbin_pad_added_cb(GstElement *webrtcbin, GstPad *new_pad, gpo
 	webrtc_s *webrtc = (webrtc_s *)user_data;
 	gchar *media_type = NULL;
 	bool is_audio;
+	g_autoptr(GMutexLocker) locker = NULL;
 
 	RET_IF(webrtc == NULL, "webrtc is NULL");
 
 	if (GST_PAD_DIRECTION(new_pad) != GST_PAD_SRC)
 		return;
 
+	locker = g_mutex_locker_new(&webrtc->mutex);
+
 	RET_IF(webrtc->pend_state == WEBRTC_STATE_IDLE, "pend_state is IDLE, skip it");
 
 	LOG_INFO("new pad[%s] is added", GST_PAD_NAME(new_pad));
@@ -1473,6 +1476,9 @@ static void __webrtcbin_pad_added_cb(GstElement *webrtcbin, GstPad *new_pad, gpo
 	is_audio = _is_audio_media_type(media_type);
 	g_free(media_type);
 
+	/* NOTE: Unlock the mutex here before calling track added user callback */
+	g_clear_pointer(&locker, g_mutex_locker_free);
+
 	if ((!is_audio && webrtc->encoded_video_frame_cb.callback) ||
 		(is_audio && webrtc->encoded_audio_frame_cb.callback)) {
 		ret = _add_forwarding_sink_bin(webrtc, new_pad, is_audio);
diff --git a/src/webrtc_stats.c b/src/webrtc_stats.c
index ffc95007..7489a023 100644
--- a/src/webrtc_stats.c
+++ b/src/webrtc_stats.c
@@ -973,6 +973,11 @@ void _webrtcbin_foreach_stats(webrtc_s *webrtc, int type_mask, void *callback, v
 static gboolean __get_stats_periodically(gpointer user_data)
 {
 	webrtc_s *webrtc = (webrtc_s *)user_data;
+	g_autoptr(GMutexLocker) locker = NULL;
+
+	RET_VAL_IF(webrtc == NULL, G_SOURCE_REMOVE, "webrtc is NULL");
+
+	locker = g_mutex_locker_new(&webrtc->mutex);
 
 	if (webrtc->state == WEBRTC_STATE_PLAYING)
 		_webrtcbin_foreach_stats(webrtc, WEBRTC_STATS_TYPE_GST_ALL, NULL, NULL);