From: Sangchul Lee <sc11.lee@samsung.com>
Date: Thu, 8 Apr 2021 08:17:15 +0000 (+0900)
Subject: webrtc_test: Add missing gst_buffer_unref()
X-Git-Tag: submit/tizen/20210729.023123~97
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=ffe701fced4e379aca06e2735f84a68f48e85ab4;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_test: Add missing gst_buffer_unref()

[Version] 0.1.144
[Issue Type] Test application

Change-Id: I3e6b69854157ec5743475d6422605295dc26f1f5
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index d61aa7aa..1e1efc0c 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:    0.1.143
+Version:    0.1.144
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/test/webrtc_test.c b/test/webrtc_test.c
index 005eac47..644f6a5b 100644
--- a/test/webrtc_test.c
+++ b/test/webrtc_test.c
@@ -1845,7 +1845,7 @@ static int __fill_gst_buffer_mapped_data_from_packet(GstBuffer *buffer, media_pa
 	return 0;
 }
 
-GstBuffer *__get_buffer_from_packet(media_packet_h packet)
+GstBuffer *__alloc_buffer_from_packet(media_packet_h packet)
 {
 	int ret;
 	guchar *data_ptr;
@@ -1854,13 +1854,6 @@ GstBuffer *__get_buffer_from_packet(media_packet_h packet)
 	guint64 dts = 0;
 	guint64 duration = 0;
 	guint64 size = 0;
-	GstBuffer *buffer = NULL;
-
-	media_packet_get_extra(packet, (void **)&buffer);
-	if (buffer) {
-		g_print("external gstbuffer[%p]\n", buffer);
-		return buffer;
-	}
 
 	media_packet_get_buffer_data_ptr(packet, (void **)&data_ptr);
 	if (data_ptr == NULL) {
@@ -1933,17 +1926,25 @@ static void __stream_handoff_cb(GstElement *object, GstBuffer *buffer, GstPad *p
 	}
 
 #ifdef __DEBUG_VALIDATE_MEDIA_PACKET_SOURCE__
-	GstBuffer *buffer_from_packet = __get_buffer_from_packet(packet);
-	if (!buffer_from_packet) {
-		media_packet_destroy(packet);
-		g_printerr("buffer_from_packet is NULL\n");
-		return;
+	GstBuffer *buffer_from_packet = NULL;
+	bool is_new_buffer = false;
+	media_packet_get_extra(packet, (void **)&buffer_from_packet);
+	if (buffer_from_packet) {
+		g_print("external gstbuffer[%p]\n", buffer_from_packet);
+	} else {
+		buffer_from_packet = __alloc_buffer_from_packet(packet);
+		if (!buffer_from_packet) {
+			media_packet_destroy(packet);
+			return;
+		}
+		is_new_buffer = true;
 	}
-
 	GstFlowReturn gst_ret = GST_FLOW_OK;
 	g_signal_emit_by_name(G_OBJECT(packet_source->appsrc), "push-buffer", buffer_from_packet, &gst_ret, NULL);
 	if (gst_ret != GST_FLOW_OK)
 		g_printerr("failed to 'push-buffer', gst_ret[0x%x]\n", gst_ret);
+	if (is_new_buffer)
+		gst_buffer_unref(buffer_from_packet);
 #else
 	if (webrtc_media_packet_source_push_packet(packet_source->webrtc, packet_source->source_id, packet) != WEBRTC_ERROR_NONE)
 		g_printerr("failed to webrtc_media_packet_source_push_packet()\n");