From: Sangchul Lee <sc11.lee@samsung.com>
Date: Fri, 1 Apr 2022 01:19:28 +0000 (+0900)
Subject: Fix spacing
X-Git-Tag: submit/tizen/20220404.065455^0
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F26%2F273226%2F3;p=platform%2Fcore%2Fapi%2Fwebrtc.git

Fix spacing

[Version] 0.3.80
[Issue Type] Coding convention

Change-Id: Idbb43d9e817afe715c0ca1d9c956171c262d61ed
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 504bc4e3..a0c833b8 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:    0.3.79
+Version:    0.3.80
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index 872fecd4..38079bab 100644
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -1387,7 +1387,7 @@ static void __webrtcbin_pad_added_cb(GstElement *webrtcbin, GstPad *new_pad, gpo
 	media_type = _get_media_type_from_pad(new_pad);
 	RET_IF(media_type == NULL, "media_type is NULL");
 
-	if(!_is_supported_media_type(media_type)) {
+	if (!_is_supported_media_type(media_type)) {
 		g_free(media_type);
 		return;
 	}
diff --git a/src/webrtc_sink.c b/src/webrtc_sink.c
index ad1a7bfd..be983f55 100644
--- a/src/webrtc_sink.c
+++ b/src/webrtc_sink.c
@@ -387,7 +387,7 @@ static void __decodebin_pad_added_cb(GstElement *decodebin, GstPad *new_pad, gpo
 
 	media_type = _get_mime_type_from_pad(new_pad);
 
-	if(!_is_supported_media_type(media_type)) {
+	if (!_is_supported_media_type(media_type)) {
 		g_free(media_type);
 		return;
 	}
diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index c8679783..81779e53 100644
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -1543,7 +1543,7 @@ static int __build_audiosrc(webrtc_s *webrtc, webrtc_gst_slot_s *source, bool us
 	source->zerocopy_enabled = __is_hw_encoder_used(webrtc, source->type, source->media_types);
 
 	source_factory_name = __get_source_element(webrtc, use_mic ? WEBRTC_MEDIA_SOURCE_TYPE_MIC : WEBRTC_MEDIA_SOURCE_TYPE_AUDIOTEST);
-	if (!(audiosrc = _create_element(source_factory_name, use_mic ? ELEMENT_NAME_MIC_SRC: NULL)))
+	if (!(audiosrc = _create_element(source_factory_name, use_mic ? ELEMENT_NAME_MIC_SRC : NULL)))
 		return WEBRTC_ERROR_INVALID_OPERATION;
 	APPEND_ELEMENT(element_list, audiosrc);
 
@@ -1945,7 +1945,7 @@ static GstPadProbeReturn __fakesink_block_probe_cb(GstPad *pad,  GstPadProbeInfo
 static GstPadProbeReturn __fakesink_probe_cb(GstPad *pad,  GstPadProbeInfo *info, gpointer u_data)
 {
 	webrtc_gst_slot_s *source = u_data;
-	GstCaps * caps= NULL;
+	GstCaps *caps = NULL;
 	GstElement *appsrc = NULL;
 	gchar *media_type = NULL;
 	int av_idx;
@@ -1953,7 +1953,7 @@ static GstPadProbeReturn __fakesink_probe_cb(GstPad *pad,  GstPadProbeInfo *info
 	media_type = _get_media_type_from_pad(pad);
 	RET_VAL_IF(media_type == NULL, GST_PAD_PROBE_OK, "media_type is NULL");
 
-	if(!_is_supported_media_type(media_type)) {
+	if (!_is_supported_media_type(media_type)) {
 		g_free(media_type);
 		return GST_PAD_PROBE_OK;
 	}
@@ -2153,7 +2153,7 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
 	media_type = _get_mime_type_from_pad(pad);
 	RET_IF(media_type == NULL, "media_type is NULL");
 
-	if(!_is_supported_media_type(media_type)) {
+	if (!_is_supported_media_type(media_type)) {
 		g_free(media_type);
 		return;
 	}
@@ -4135,7 +4135,7 @@ static void __loopback_decodebin_pad_added_cb(GstElement *decodebin, GstPad *new
 	media_type = _get_mime_type_from_pad(new_pad);
 	LOG_INFO("source_id[%u], media_type[%s], new_pad[%s]", source->id, media_type, GST_PAD_NAME(new_pad));
 
-	if(!_is_supported_media_type(media_type)) {
+	if (!_is_supported_media_type(media_type)) {
 		g_free(media_type);
 		return;
 	}
@@ -4615,7 +4615,7 @@ int _set_filesrc_looping(webrtc_s *webrtc, unsigned int source_id, bool looping)
 	return WEBRTC_ERROR_NONE;
 }
 
-int _get_filesrc_looping(webrtc_s * webrtc, unsigned int source_id, bool *looping)
+int _get_filesrc_looping(webrtc_s *webrtc, unsigned int source_id, bool *looping)
 {
 	const webrtc_gst_slot_s *source;
 
diff --git a/test/webrtc_test.c b/test/webrtc_test.c
index 2bca3cce..0b27b57b 100644
--- a/test/webrtc_test.c
+++ b/test/webrtc_test.c
@@ -1241,7 +1241,7 @@ static void _webrtc_data_channel_send_file(int index, const char *file_path)
 		g_free(expected_name);
 
 		sum_size = 0;
-		while((read_size = read(fd, buffer, BUFFER_SIZE)) > 0) {
+		while ((read_size = read(fd, buffer, BUFFER_SIZE)) > 0) {
 			sum_size += read_size;
 			g_print("%s : [%llu / %llu]bytes is read.", file_path, sum_size, (unsigned long long)st.st_size);