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 504bc4e3a58d79ec1d4dde2ad71bfe88df71b305..a0c833b892322cc60d356f9c7221ea4ed2e43cbf 100644 (file)
--- 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 872fecd407e0d5c85d7723c3ab9902e99a99655e..38079bab1524f52759c25ad7da170a65064fa7c6 100644 (file)
--- 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 ad1a7bfd00fd2a5a54ef8087fe9e69fab121025b..be983f5562bca7221d5999647fbf8c0a9912e0d1 100644 (file)
--- 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 c8679783ccc267f8543af4eea3994fa9cc263980..81779e53e66cc112872e2b8f706004f9bd2afc7d 100644 (file)
--- 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 2bca3cce4a0f0949afacda5aa3440f0edf598dfa..0b27b57b7bfbfea5975b870bfce1829c88f04a38 100644 (file)
--- 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);