From: backto.kim <backto.kim@samsung.com>
Date: Wed, 6 Oct 2021 07:30:28 +0000 (+0900)
Subject: webrtc_source: rearrange codes of __create_rest_of_elements_for_filesrc_pipeline... 
X-Git-Tag: submit/tizen/20211020.012736~15
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F66%2F264966%2F4;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_source: rearrange codes of __create_rest_of_elements_for_filesrc_pipeline() to reduce code complexity

[Version] 0.2.115
[Issue Type] Refactoring

Change-Id: Ibed6a8d5b613e91af1a8a6c6f944e59dcdeae569
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index cf151355..6fbc5c70 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:    0.2.114
+Version:    0.2.115
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index ee5ebb22..915382de 100644
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -1775,28 +1775,21 @@ static GstPadProbeReturn __fakesink_probe_cb(GstPad *pad,  GstPadProbeInfo *info
 	return GST_PAD_PROBE_REMOVE;
 }
 
-static int __create_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *source, GstElement *payloader, bool is_audio)
+static GstElement * __prepare_capsfilter_for_filesrc_pipeline(webrtc_gst_slot_s *source, bool is_audio)
 {
-	GstBin *bin = NULL;
 	GstElement *capsfilter = NULL;
-	GstElement *fakesink = NULL;
 	GstCaps *sink_caps = NULL;
-	unsigned int payload_id;
-	GstPad *sink_pad = NULL;
+	unsigned int payload_id = 0;
 
-	RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
-	RET_VAL_IF(payloader == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "payloader is NULL");
-	RET_VAL_IF(source->filesrc_pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "filesrc_pipeline is NULL");
-
-	bin = GST_BIN(source->filesrc_pipeline);
+	RET_VAL_IF(source == NULL, NULL, "source is NULL");
 
 	if (!(capsfilter = _create_element(DEFAULT_ELEMENT_CAPSFILTER, is_audio ? DEFAULT_NAME_AUDIO_CAPSFILTER : DEFAULT_NAME_VIDEO_CAPSFILTER)))
-		return WEBRTC_ERROR_INVALID_OPERATION;
+		return NULL;
 
 	payload_id = __get_available_payload_id(source->webrtc);
 	if (payload_id == 0) {
 		SAFE_GST_OBJECT_UNREF(capsfilter);
-		return WEBRTC_ERROR_INVALID_OPERATION;
+		return NULL;
 	}
 
 	source->av[is_audio ? AV_IDX_AUDIO : AV_IDX_VIDEO].payload_id = payload_id;
@@ -1806,40 +1799,75 @@ static int __create_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *sou
 		gst_caps_unref(sink_caps);
 	}
 
-	if (!(fakesink = _create_element("fakesink", is_audio ? DEFAULT_NAME_AUDIO_FAKESINK : DEFAULT_NAME_VIDEO_FAKESINK))) {
-		SAFE_GST_OBJECT_UNREF(capsfilter);
-		return WEBRTC_ERROR_INVALID_OPERATION;
-	}
+	return capsfilter;
+}
+
+static GstElement * __prepare_fakesink_for_filesrc_pipeline(webrtc_gst_slot_s *source, bool is_audio)
+{
+	GstElement *fakesink = NULL;
+	GstPad *sink_pad = NULL;
+
+	RET_VAL_IF(source == NULL, NULL, "source is NULL");
+
+	if (!(fakesink = _create_element("fakesink", is_audio ? DEFAULT_NAME_AUDIO_FAKESINK : DEFAULT_NAME_VIDEO_FAKESINK)))
+		return NULL;
 
 	sink_pad = gst_element_get_static_pad(fakesink, "sink");
 	gst_pad_add_probe(sink_pad, GST_PAD_PROBE_TYPE_BUFFER, __fakesink_probe_cb, source, NULL);
 	gst_object_unref(sink_pad);
 
-	g_object_set(G_OBJECT(fakesink), "sync", true, NULL);
-	g_object_set(fakesink, "signal-handoffs", TRUE, NULL);
+	g_object_set(G_OBJECT(fakesink),
+		"sync", TRUE,
+		"signal-handoffs", TRUE,
+		NULL);
+
 	g_signal_connect(fakesink, "handoff", is_audio ? G_CALLBACK(__filesrc_pipeline_audio_stream_handoff_cb) : G_CALLBACK(__filesrc_pipeline_video_stream_handoff_cb), (gpointer)source);
 
-	gst_bin_add_many(GST_BIN(source->filesrc_pipeline), capsfilter, fakesink, NULL);
+	return fakesink;
+}
+
+static int __create_rest_of_elements_for_filesrc_pipeline(webrtc_gst_slot_s *source, GstElement *payloader, bool is_audio)
+{
+	GstBin *bin = NULL;
+	GstElement *capsfilter = NULL;
+	GstElement *fakesink = NULL;
+
+	RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
+	RET_VAL_IF(payloader == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "payloader is NULL");
+	RET_VAL_IF(source->filesrc_pipeline == NULL, WEBRTC_ERROR_INVALID_OPERATION, "filesrc_pipeline is NULL");
+
+	bin = GST_BIN(source->filesrc_pipeline);
+
+	if (!(capsfilter = __prepare_capsfilter_for_filesrc_pipeline(source, is_audio)))
+		return WEBRTC_ERROR_INVALID_OPERATION;
+
+	if (!(fakesink = __prepare_fakesink_for_filesrc_pipeline(source, is_audio))) {
+		SAFE_GST_OBJECT_UNREF(capsfilter);
+		return WEBRTC_ERROR_INVALID_OPERATION;
+	}
+
+	gst_bin_add_many(bin, capsfilter, fakesink, NULL);
 
 	if (!gst_element_link_many(payloader, capsfilter, fakesink, NULL)) {
 		LOG_ERROR("failed to gst_element_link_many()");
-		gst_bin_remove_many(bin, capsfilter, fakesink, NULL);
-		return WEBRTC_ERROR_INVALID_OPERATION;
+		goto error;
 	}
 
 	if (!gst_element_sync_state_with_parent(capsfilter)) {
 		LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(capsfilter));
-		gst_bin_remove_many(bin, capsfilter, fakesink, NULL);
-		return WEBRTC_ERROR_INVALID_OPERATION;
+		goto error;
 	}
 
 	if (!gst_element_sync_state_with_parent(fakesink)) {
 		LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(fakesink));
-		gst_bin_remove_many(bin, capsfilter, fakesink, NULL);
-		return WEBRTC_ERROR_INVALID_OPERATION;
+		goto error;
 	}
 
 	return WEBRTC_ERROR_NONE;
+
+error:
+	gst_bin_remove_many(bin, capsfilter, fakesink, NULL);
+	return WEBRTC_ERROR_INVALID_OPERATION;
 }
 
 static GstElement * __link_decodebin_with_payload(GstPad *pad, webrtc_gst_slot_s *source, bool is_audio, bool create)