From: Sangchul Lee <sc11.lee@samsung.com>
Date: Fri, 28 Aug 2020 08:45:44 +0000 (+0900)
Subject: Add webrtcbin element to pipeline and add _gst_destroy_pipeline() sub-function
X-Git-Tag: submit/tizen/20210729.023123~240
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=711ae728016b36b2cbe2a137783a04214e666122;p=platform%2Fcore%2Fapi%2Fwebrtc.git

Add webrtcbin element to pipeline and add _gst_destroy_pipeline() sub-function

[Version] 0.1.2
[Issue Type] Improvement

Change-Id: I324a02b04453eb9dfebe4105241e55036ee14c64
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index 6de5cf9f..5a7fc244 100644
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -96,6 +96,7 @@ do { \
 } while (0)
 
 #define SAFE_FREE(src)    { if (src) { free(src); src = NULL; } }
+#define SAFE_STR(str)     (str) ? str : "null"
 
 #define SOURCES_MAX    8
 
@@ -105,12 +106,13 @@ typedef struct _webrtc_ini_s {
 } webrtc_ini_s;
 
 typedef struct _webrtc_gst_src_s {
-	int source_id;
+	unsigned int source_id;
 	GstElement *bin;
 } webrtc_gst_src_s;
 
 typedef struct _webrtc_gst_s {
 	GstElement *pipeline;
+	GstElement *webrtcbin;
 	GstBus *bus;
 	guint bus_watcher;
 	webrtc_gst_src_s *audiosrc[SOURCES_MAX];
@@ -131,6 +133,7 @@ typedef struct _webrtc_s {
 int _ini_load(webrtc_s *webrtc);
 int _gst_init(webrtc_s *webrtc);
 int _gst_build_pipeline(webrtc_s *webrtc);
+void _gst_destroy_pipeline(webrtc_s *webrtc);
 
 #ifdef __cplusplus
 }
diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 701f3b87..95e1e243 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:    0.1.1
+Version:    0.1.2
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc.c b/src/webrtc.c
index 1a624711..6ee3ce48 100644
--- a/src/webrtc.c
+++ b/src/webrtc.c
@@ -58,10 +58,7 @@ int webrtc_destroy(webrtc_h webrtc)
 
 	/* set state to null here */
 
-	if (_webrtc->gst.bus)
-		gst_object_unref(_webrtc->gst.bus);
-	if (_webrtc->gst.pipeline)
-		gst_object_unref(_webrtc->gst.pipeline);
+	_gst_destroy_pipeline(_webrtc);
 
 	g_mutex_unlock(&_webrtc->mutex);
 	g_mutex_clear(&_webrtc->mutex);
diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index 2c1dc30b..8b400d12 100644
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -80,6 +80,20 @@ static gboolean __bus_watch_cb(GstBus *bus, GstMessage *message, gpointer user_d
 	return TRUE;
 }
 
+static GstElement *__element_create(const char *factory_name, const char *name)
+{
+	GstElement *element = NULL;
+
+	RET_VAL_IF(!factory_name, NULL, "factory name is NULL");
+
+	element = gst_element_factory_make(factory_name, name);
+	RET_VAL_IF(!element, NULL, "element is NULL [%s]", factory_name);
+
+	LOG_DEBUG("created element [%s, %s]", factory_name, SAFE_STR(name));
+
+	return element;
+}
+
 int _ini_load(webrtc_s *webrtc)
 {
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
@@ -142,19 +156,58 @@ int _gst_init(webrtc_s *webrtc)
 
 int _gst_build_pipeline(webrtc_s *webrtc)
 {
-	int ret = WEBRTC_ERROR_NONE;
-
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 
 	webrtc->gst.pipeline = gst_pipeline_new("native-webrtc-pipeline");
 	RET_VAL_IF(webrtc->gst.pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "pipeline is NULL");
 
-	webrtc->gst.bus = gst_pipeline_get_bus(GST_PIPELINE(webrtc->gst.pipeline));
-	RET_VAL_IF(webrtc->gst.bus == NULL, WEBRTC_ERROR_INVALID_OPERATION, "bus is NULL");
+	if (!(webrtc->gst.bus = gst_pipeline_get_bus(GST_PIPELINE(webrtc->gst.pipeline)))) {
+		LOG_ERROR("failed to gst_pipeline_get_bus()");
+		goto error;
+	}
+
+	if ((webrtc->gst.bus_watcher = gst_bus_add_watch(webrtc->gst.bus, (GstBusFunc) __bus_watch_cb, webrtc)) == 0) {
+		LOG_ERROR("failed to gst_bus_add_watch()");
+		goto error;
+	}
+
+	if (!(webrtc->gst.webrtcbin = __element_create("webrtcbin", NULL))) {
+		LOG_ERROR("failed to create webrtcbin");
+		goto error;
+	}
+
+	if (!gst_bin_add(GST_BIN(webrtc->gst.pipeline), webrtc->gst.webrtcbin)) {
+		LOG_ERROR("failed to gst_bin_add(), [%s] -> [%s] pipeline", GST_ELEMENT_NAME(webrtc->gst.webrtcbin), GST_ELEMENT_NAME(webrtc->gst.pipeline));
+		goto error;
+	}
 
-	webrtc->gst.bus_watcher = gst_bus_add_watch(webrtc->gst.bus, (GstBusFunc) __bus_watch_cb, webrtc);
+	return WEBRTC_ERROR_NONE;
+
+error:
+	_gst_destroy_pipeline(webrtc);
+	return WEBRTC_ERROR_INVALID_OPERATION;
+}
 
-	/* FIXME : initial bin setting */
+void _gst_destroy_pipeline(webrtc_s *webrtc)
+{
+	if (!webrtc)
+		return;
 
-	return ret;
+	if (webrtc->gst.bus_watcher > 0) {
+		gst_bus_remove_watch(webrtc->gst.bus);
+		webrtc->gst.bus_watcher = 0;
+	}
+	if (webrtc->gst.bus) {
+		gst_object_unref(webrtc->gst.bus);
+		webrtc->gst.bus = NULL;
+	}
+	if (webrtc->gst.webrtcbin) {
+		gst_bin_remove(GST_BIN(webrtc->gst.pipeline), webrtc->gst.webrtcbin);
+		webrtc->gst.webrtcbin = NULL;
+	}
+	if (webrtc->gst.pipeline) {
+		gst_object_unref(webrtc->gst.pipeline);
+		webrtc->gst.pipeline = NULL;
+	}
 }
+