From: Sangchul Lee <sc11.lee@samsung.com>
Date: Wed, 21 Apr 2021 11:52:06 +0000 (+0900)
Subject: webrtc_private: Destroy sink pipeline in case of webrtc_stop()
X-Git-Tag: submit/tizen/20210729.023123~91
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F42%2F257242%2F2;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_private: Destroy sink pipeline in case of webrtc_stop()

When the webrtc_stop() is called, the state is changed to IDLE.
The state will be changed to NEGOTIATING with following webrtc_start()
again which means the re-negotiation is required.

Therefore, releasing the previous sink pipeline should be performed
inside of webrtc_stop().

[Version] 0.1.150
[Issue Type] Improvement

Change-Id: I1bd03ecf8b8f27d93dec7d1488fa20223299a737
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index f2a874d2..9b46ed9d 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:    0.1.149
+Version:    0.1.150
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index 4e42723a..0da26394 100644
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -1035,23 +1035,36 @@ error:
 	return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
-void _gst_destroy_pipeline(webrtc_s *webrtc)
+static void __destroy_sink_pipeline(webrtc_s *webrtc)
 {
-	if (!webrtc)
-		return;
+	RET_IF(webrtc == NULL, "webrtc is NULL");
 
-	if (webrtc->track_build_contexts) {
-		g_hash_table_destroy(webrtc->track_build_contexts);
-		webrtc->track_build_contexts = NULL;
-	}
 	if (webrtc->gst.sink_slots) {
 		g_hash_table_destroy(webrtc->gst.sink_slots);
 		webrtc->gst.sink_slots = NULL;
 	}
+}
+
+static void __destroy_source_pipeline(webrtc_s *webrtc)
+{
+	RET_IF(webrtc == NULL, "webrtc is NULL");
+
 	if (webrtc->gst.source_slots) {
 		g_hash_table_destroy(webrtc->gst.source_slots);
 		webrtc->gst.source_slots = NULL;
 	}
+}
+
+void _gst_destroy_pipeline(webrtc_s *webrtc)
+{
+	RET_IF(webrtc == NULL, "webrtc is NULL");
+
+	if (webrtc->track_build_contexts) {
+		g_hash_table_destroy(webrtc->track_build_contexts);
+		webrtc->track_build_contexts = NULL;
+	}
+	__destroy_sink_pipeline(webrtc);
+	__destroy_source_pipeline(webrtc);
 	if (webrtc->gst.bus_watcher > 0) {
 		gst_bus_remove_watch(webrtc->gst.bus);
 		webrtc->gst.bus_watcher = 0;
@@ -1556,6 +1569,9 @@ int _webrtc_stop(webrtc_s *webrtc)
 	ret = _release_all_resources(webrtc);
 	RET_VAL_WITH_UNLOCK_IF(ret != WEBRTC_ERROR_NONE, ret, &webrtc->mutex, "failed to release all resources");
 #endif
+
+	__destroy_sink_pipeline(webrtc);
+
 	__post_state_cb_in_idle(webrtc, WEBRTC_STATE_IDLE);
 
 	LOG_INFO("webrtc[%p] is stopped", webrtc);