From: Sangchul Lee <sc11.lee@samsung.com>
Date: Thu, 22 Apr 2021 04:36:10 +0000 (+0900)
Subject: webrtc_source: Release gst resources in case of error
X-Git-Tag: submit/tizen/20210729.023123~89
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=fad5b007d8b0a47ba15f88b09bfa086b3fc2160e;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_source: Release gst resources in case of error

[Version] 0.1.152
[Issue Type] Improvement

Change-Id: Ic9a5e8ffadae1384e5d3e2b7787e2df29382fabe
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index 0e24579e..8fbef603 100644
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -159,6 +159,7 @@ do { \
 } while (0)
 
 #define SAFE_FREE(src)    { if (src) { free(src); src = NULL; } }
+#define SAFE_GST_OBJECT_UNREF(src)    { if (src) { gst_object_unref(src); src = NULL; } }
 #define SAFE_STR(str)     (str) ? str : "null"
 
 #define CREATE_ELEMENT_FROM_REGISTRY(x_elem_info, x_klass_name, x_sink_caps, x_src_caps, x_element) \
diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index e7e83341..ff14f56f 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:    0.1.151
+Version:    0.1.152
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index fc63e22d..907a552d 100644
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -711,11 +711,11 @@ static int __build_camerasrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 {
 	int ret = WEBRTC_ERROR_NONE;
 	GstElement *camerasrc;
-	GstElement *capsfilter;
-	GstElement *videoenc;
-	GstElement *videopay;
-	GstElement *queue;
-	GstElement *capsfilter2;
+	GstElement *capsfilter = NULL;
+	GstElement *videoenc = NULL;
+	GstElement *videopay = NULL;
+	GstElement *queue = NULL;
+	GstElement *capsfilter2 = NULL;
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
@@ -738,26 +738,44 @@ static int __build_camerasrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 		g_object_set(G_OBJECT(camerasrc), "empty-buffer-timeout", 0, NULL);
 
 	if ((ret = __create_rest_of_elements(webrtc, source, &capsfilter, &videoenc, &videopay, &queue, &capsfilter2)) != WEBRTC_ERROR_NONE)
-		return ret;
+		goto exit;
 
 	gst_bin_add_many(source->bin, camerasrc, capsfilter, videoenc, videopay, queue, capsfilter2, NULL);
 	if (!gst_element_link_many(camerasrc, capsfilter, videoenc, videopay, queue, capsfilter2, NULL)) {
 		LOG_ERROR("failed to gst_element_link_many()");
-		return WEBRTC_ERROR_INVALID_OPERATION;
+		ret = WEBRTC_ERROR_INVALID_OPERATION;
+		goto exit_with_remove_from_bin;
 	}
 
-	return _set_ghost_pad_target(source->src_pad, capsfilter2, true);
+	ret = _set_ghost_pad_target(source->src_pad, capsfilter2, true);
+	if (ret != WEBRTC_ERROR_NONE)
+		goto exit_with_remove_from_bin;
+
+	return WEBRTC_ERROR_NONE;
+
+exit_with_remove_from_bin:
+	/* elements will be dereferenced */
+	gst_bin_remove_many(source->bin, camerasrc, capsfilter, videoenc, videopay, queue, capsfilter2, NULL);
+	return ret;
+exit:
+	SAFE_GST_OBJECT_UNREF(camerasrc);
+	SAFE_GST_OBJECT_UNREF(capsfilter);
+	SAFE_GST_OBJECT_UNREF(videoenc);
+	SAFE_GST_OBJECT_UNREF(videopay);
+	SAFE_GST_OBJECT_UNREF(queue);
+	SAFE_GST_OBJECT_UNREF(capsfilter2);
+	return ret;
 }
 
 static int __build_audiosrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 {
 	int ret = WEBRTC_ERROR_NONE;
 	GstElement *audiosrc;
-	GstElement *capsfilter;
-	GstElement *audioenc;
-	GstElement *audiopay;
-	GstElement *queue;
-	GstElement *capsfilter2;
+	GstElement *capsfilter = NULL;
+	GstElement *audioenc = NULL;
+	GstElement *audiopay = NULL;
+	GstElement *queue = NULL;
+	GstElement *capsfilter2 = NULL;
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
@@ -772,26 +790,44 @@ static int __build_audiosrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 	}
 
 	if ((ret = __create_rest_of_elements(webrtc, source, &capsfilter, &audioenc, &audiopay, &queue, &capsfilter2)) != WEBRTC_ERROR_NONE)
-		return ret;
+		goto exit;
 
 	gst_bin_add_many(source->bin, audiosrc, capsfilter, audioenc, audiopay, queue, capsfilter2, NULL);
 	if (!gst_element_link_many(audiosrc, capsfilter, audioenc, audiopay, queue, capsfilter2, NULL)) {
 		LOG_ERROR("failed to gst_element_link_many()");
-		return WEBRTC_ERROR_INVALID_OPERATION;
+		ret = WEBRTC_ERROR_INVALID_OPERATION;
+		goto exit_with_remove_from_bin;
 	}
 
-	return _set_ghost_pad_target(source->src_pad, capsfilter2, true);
+	ret = _set_ghost_pad_target(source->src_pad, capsfilter2, true);
+	if (ret != WEBRTC_ERROR_NONE)
+		goto exit_with_remove_from_bin;
+
+	return WEBRTC_ERROR_NONE;
+
+exit_with_remove_from_bin:
+	/* elements will be dereferenced */
+	gst_bin_remove_many(source->bin, audiosrc, capsfilter, audioenc, audiopay, queue, capsfilter2, NULL);
+	return ret;
+exit:
+	SAFE_GST_OBJECT_UNREF(audiosrc);
+	SAFE_GST_OBJECT_UNREF(capsfilter);
+	SAFE_GST_OBJECT_UNREF(audioenc);
+	SAFE_GST_OBJECT_UNREF(audiopay);
+	SAFE_GST_OBJECT_UNREF(queue);
+	SAFE_GST_OBJECT_UNREF(capsfilter2);
+	return ret;
 }
 
 static int __build_videotestsrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 {
 	int ret = WEBRTC_ERROR_NONE;
 	GstElement *videotestsrc;
-	GstElement *capsfilter;
-	GstElement *videoenc;
-	GstElement *videopay;
-	GstElement *queue;
-	GstElement *capsfilter2;
+	GstElement *capsfilter = NULL;
+	GstElement *videoenc = NULL;
+	GstElement *videopay = NULL;
+	GstElement *queue = NULL;
+	GstElement *capsfilter2 = NULL;
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
@@ -807,26 +843,44 @@ static int __build_videotestsrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 	g_object_set(G_OBJECT(videotestsrc), "is-live", TRUE, "pattern", 18, NULL); /* 18: ball */
 
 	if ((ret = __create_rest_of_elements(webrtc, source, &capsfilter, &videoenc, &videopay, &queue, &capsfilter2)) != WEBRTC_ERROR_NONE)
-		return ret;
+		goto exit;
 
 	gst_bin_add_many(source->bin, videotestsrc, capsfilter, videoenc, videopay, queue, capsfilter2, NULL);
 	if (!gst_element_link_many(videotestsrc, capsfilter, videoenc, videopay, queue, capsfilter2, NULL)) {
 		LOG_ERROR("failed to gst_element_link_many()");
-		return WEBRTC_ERROR_INVALID_OPERATION;
+		ret = WEBRTC_ERROR_INVALID_OPERATION;
+		goto exit_with_remove_from_bin;
 	}
 
-	return _set_ghost_pad_target(source->src_pad, capsfilter2, true);
+	ret = _set_ghost_pad_target(source->src_pad, capsfilter2, true);
+	if (ret != WEBRTC_ERROR_NONE)
+		goto exit_with_remove_from_bin;
+
+	return WEBRTC_ERROR_NONE;
+
+exit_with_remove_from_bin:
+	/* elements will be dereferenced */
+	gst_bin_remove_many(source->bin, videotestsrc, capsfilter, videoenc, videopay, queue, capsfilter2, NULL);
+	return ret;
+exit:
+	SAFE_GST_OBJECT_UNREF(videotestsrc);
+	SAFE_GST_OBJECT_UNREF(capsfilter);
+	SAFE_GST_OBJECT_UNREF(videoenc);
+	SAFE_GST_OBJECT_UNREF(videopay);
+	SAFE_GST_OBJECT_UNREF(queue);
+	SAFE_GST_OBJECT_UNREF(capsfilter2);
+	return ret;
 }
 
 static int __build_audiotestsrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 {
 	int ret = WEBRTC_ERROR_NONE;
 	GstElement *audiotestsrc;
-	GstElement *capsfilter;
-	GstElement *audioenc;
-	GstElement *audiopay;
-	GstElement *queue;
-	GstElement *capsfilter2;
+	GstElement *capsfilter = NULL;
+	GstElement *audioenc = NULL;
+	GstElement *audiopay = NULL;
+	GstElement *queue = NULL;
+	GstElement *capsfilter2 = NULL;
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
@@ -842,15 +896,33 @@ static int __build_audiotestsrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
 	g_object_set(G_OBJECT(audiotestsrc), "is-live", TRUE, NULL);
 
 	if ((ret = __create_rest_of_elements(webrtc, source, &capsfilter, &audioenc, &audiopay, &queue, &capsfilter2)) != WEBRTC_ERROR_NONE)
-		return ret;
+		goto exit;
 
 	gst_bin_add_many(source->bin, audiotestsrc, capsfilter, audioenc, audiopay, queue, capsfilter2, NULL);
 	if (!gst_element_link_many(audiotestsrc, capsfilter, audioenc, audiopay, queue, capsfilter2, NULL)) {
 		LOG_ERROR("failed to gst_element_link_many()");
-		return WEBRTC_ERROR_INVALID_OPERATION;
+		ret = WEBRTC_ERROR_INVALID_OPERATION;
+		goto exit_with_remove_from_bin;
 	}
 
-	return _set_ghost_pad_target(source->src_pad, capsfilter2, true);
+	ret = _set_ghost_pad_target(source->src_pad, capsfilter2, true);
+	if (ret != WEBRTC_ERROR_NONE)
+		goto exit_with_remove_from_bin;
+
+	return WEBRTC_ERROR_NONE;
+
+exit_with_remove_from_bin:
+	/* elements will be dereferenced */
+	gst_bin_remove_many(source->bin, audiotestsrc, capsfilter, audioenc, audiopay, queue, capsfilter2, NULL);
+	return ret;
+exit:
+	SAFE_GST_OBJECT_UNREF(audiotestsrc);
+	SAFE_GST_OBJECT_UNREF(capsfilter);
+	SAFE_GST_OBJECT_UNREF(audioenc);
+	SAFE_GST_OBJECT_UNREF(audiopay);
+	SAFE_GST_OBJECT_UNREF(queue);
+	SAFE_GST_OBJECT_UNREF(capsfilter2);
+	return ret;
 }
 
 static void _appsrc_need_data_cb(GstElement *appsrc, guint size, gpointer data)