webrtc_source_mediapacket: apply coding rule for internal functions

Plus, move _set_mediapacketsrc_codec_info() to the proper header file webrtc_private.h

[Version] 0.3.171
[Issue Type] Convention

Change-Id: Icd2ec651ab5e3d450379c206fbf5c5c4f2515253

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index c0a798377f42caa33269d0c4bf17df5388e30e91..2f93c76ac2ef7d197bdb4a72b8c121039109e3c3 100644 (file)
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -692,6 +692,7 @@ int _complete_rest_of_mediapacketsrc(webrtc_gst_slot_s *source, GstPad **src_pad
 int _complete_mediapacketsrc_from_encoded_format(webrtc_s *webrtc, webrtc_gst_slot_s *source);
 GstCaps *_make_mediapacketsrc_raw_caps_from_media_format(webrtc_gst_slot_s *source);
 int _push_media_packet(webrtc_s *webrtc, unsigned int source_id, media_packet_h packet);
+int _set_mediapacketsrc_codec_info(webrtc_s *webrtc, webrtc_gst_slot_s *source, media_format_mimetype_e mime_type);
 
 bool _check_if_path_is_set_to_file_sources(webrtc_s *webrtc);
 int _set_rtp_packet_drop_probability(webrtc_s *webrtc, unsigned int source_id, float probability);

diff --git a/include/webrtc_source_private.h b/include/webrtc_source_private.h
index f940dfef0f45f4873ff0abe73d301cc1520f4a42..ec9016a4e3951b6f6f963d94de53c9892a1283d3 100644 (file)
--- a/include/webrtc_source_private.h
+++ b/include/webrtc_source_private.h
@@ -93,7 +93,6 @@ bool _is_encoded_format_supported(webrtc_media_source_type_e type, webrtc_ini_s
 GstCaps *_get_caps_from_encoded_audio_media_type(const char *media_type, int channels, int samplerate);
 GstCaps *_get_caps_from_encoded_video_media_type(const char *media_type, int width, int height);
 GstCaps *_make_rtp_caps(const gchar *media_type, unsigned int payload_type, webrtc_gst_slot_s *source);
-int _set_mediapacketsrc_codec_info(webrtc_s *webrtc, webrtc_gst_slot_s *source, media_format_mimetype_e mime_type);
 int _get_screen_resolution(int *width, int *height);
 const char *_get_element_name(int av_idx, gst_element_e element);
 int _set_payload_type(webrtc_s *webrtc, webrtc_gst_slot_s *source, int av_idx, const gchar *media_type);

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 202b2d1a82027a042fc51483d7189dd6364d563e..01a0b94f7d698d726e5b0fb65d448ee3ffee26b5 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:    0.3.170
+Version:    0.3.171
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source_mediapacket.c b/src/webrtc_source_mediapacket.c
index e905848a0a2b8f38345fb3b0496c782199aa36c6..e2da456738dd0a6687cd928a2111bc39c58381e3 100644 (file)
--- a/src/webrtc_source_mediapacket.c
+++ b/src/webrtc_source_mediapacket.c
@@ -208,7 +208,7 @@ static int __fill_gst_buffer_mapped_data_from_packet(GstBuffer *buffer, media_pa
 }
 
 //LCOV_EXCL_START
-static void _appsrc_need_data_cb(GstElement *appsrc, guint size, gpointer data)
+static void __appsrc_need_data_cb(GstElement *appsrc, guint size, gpointer data)
 {
        webrtc_gst_slot_s *source = (webrtc_gst_slot_s *)data;
 
@@ -228,7 +228,7 @@ static void _appsrc_need_data_cb(GstElement *appsrc, guint size, gpointer data)
        LOG_DEBUG("<<< end of the callback");
 }
 
-static void _appsrc_enough_data_cb(GstElement *appsrc, gpointer data)
+static void __appsrc_enough_data_cb(GstElement *appsrc, gpointer data)
 {
        webrtc_gst_slot_s *source = (webrtc_gst_slot_s *)data;
 
@@ -545,8 +545,8 @@ int _build_mediapacketsrc(webrtc_s *webrtc, webrtc_gst_slot_s *source)
                "format", GST_FORMAT_TIME,
                NULL);
 
-       _connect_and_append_signal(&source->signals, G_OBJECT(appsrc), "need-data", G_CALLBACK(_appsrc_need_data_cb), source);
-       _connect_and_append_signal(&source->signals, G_OBJECT(appsrc), "enough-data", G_CALLBACK(_appsrc_enough_data_cb), source);
+       _connect_and_append_signal(&source->signals, G_OBJECT(appsrc), "need-data", G_CALLBACK(__appsrc_need_data_cb), source);
+       _connect_and_append_signal(&source->signals, G_OBJECT(appsrc), "enough-data", G_CALLBACK(__appsrc_enough_data_cb), source);
 
        gst_bin_add(source->bin, appsrc);