Improve the code to check media type

[Version] 0.3.17
[Issue Type] Refactoring

Change-Id: Icc49b8e30ab2744d6d08a64f59dcc2b7c4b93c30

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index bf2022df834aa0c582ac4d3da147ab6e3d047eae..0d82d3641b864cfaf524789e73354cea085c9c08 100644 (file)
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -689,6 +689,8 @@ int _set_filesrc_looping(webrtc_s *webrtc, unsigned int source_id, bool looping)
 int _get_filesrc_looping(webrtc_s *webrtc, unsigned int source_id, bool *looping);
 int _remove_filesrc_pad_block_probe(webrtc_s *webrtc);
 
+bool _is_supported_media_type(const char *media_type);
+bool _is_audio_media_type(const char *media_type);
 #ifdef __cplusplus
 }
 #endif /* __cplusplus */

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 84b4fde453de899360fef4ff02c6c4a3ec1ade5a..cc9a9205e4dd2a48f7491c69c4499bde7835eb91 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:    0.3.16
+Version:    0.3.17
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index 27141582e46444c80b8000a2253ae10baf43b740..c3442bfefff3fa38c27fba7a6916b14879026b44 100644 (file)
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -1119,6 +1119,25 @@ static int __get_media_type_from_pad(GstPad *pad, bool *is_video)
 }
 //LCOV_EXCL_STOP
 
+bool _is_supported_media_type(const char *media_type)
+{
+       RET_VAL_IF(media_type == NULL, false, "media_type is NULL");
+
+       if (!g_strrstr(media_type, "audio") && !g_strrstr(media_type, "video")) {
+               LOG_ERROR("not supported media type [%s]", media_type);
+               return false;
+       }
+
+       return true;
+}
+
+bool _is_audio_media_type(const char *media_type)
+{
+       RET_VAL_IF(media_type == NULL, false, "media_type is NULL");
+
+       return (bool)g_strrstr(media_type, "audio");
+}
+
 static void __webrtcbin_pad_added_cb(GstElement *webrtcbin, GstPad *new_pad, gpointer user_data)
 {
        int ret = WEBRTC_ERROR_NONE;

diff --git a/src/webrtc_sink.c b/src/webrtc_sink.c
index 71348767f472f87f483546d7de52e53e1f16488f..8745e8f949d43eb4caa0cbfdf304b2fbae37fe86 100644 (file)
--- a/src/webrtc_sink.c
+++ b/src/webrtc_sink.c
@@ -370,24 +370,23 @@ static void __decodebin_pad_added_cb(GstElement *decodebin, GstPad *new_pad, gpo
                return;
 
        media_type = gst_structure_get_name(gst_caps_get_structure(gst_pad_get_current_caps(new_pad), 0));
+
+       if(!_is_supported_media_type(media_type))
+               return;
+
        LOG_INFO("decodebin[%p, name:%s] new_pad[%s] media_type[%s]", decodebin, GST_ELEMENT_NAME(decodebin), GST_PAD_NAME(new_pad), media_type);
 
        sink = __find_sink_slot(webrtc, GST_ELEMENT_NAME(decodebin));
        RET_IF(sink == NULL, "could not find an item by [%s] in sink slots", GST_ELEMENT_NAME(decodebin));
 
-       if (g_strrstr(media_type, "video")) {
-               sink->media_types |= MEDIA_TYPE_VIDEO;
-               __invoke_track_added_cb(webrtc, GST_ELEMENT_NAME(decodebin), true, true);
-               ret = __build_videosink(webrtc, decodebin, new_pad);
-
-       } else if (g_strrstr(media_type, "audio")) {
+       if (_is_audio_media_type(media_type)) {
                sink->media_types |= MEDIA_TYPE_AUDIO;
                __invoke_track_added_cb(webrtc, GST_ELEMENT_NAME(decodebin), false, true);
                ret = __build_audiosink(webrtc, decodebin, new_pad);
-
        } else {
-               LOG_ERROR("not supported media type[%s]", media_type);
-               return;
+               sink->media_types |= MEDIA_TYPE_VIDEO;
+               __invoke_track_added_cb(webrtc, GST_ELEMENT_NAME(decodebin), true, true);
+               ret = __build_videosink(webrtc, decodebin, new_pad);
        }
 
        if (ret != WEBRTC_ERROR_NONE) {

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index e82712f3dfd4c4040a5323651cd7679fc1a86a19..c047325d90d66eea0fd7b340f91056e903fc34a1 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -1899,25 +1899,23 @@ static GstPadProbeReturn __fakesink_probe_cb(GstPad *pad,  GstPadProbeInfo *info
        webrtc_gst_slot_s *source = u_data;
        GstCaps *new_cap = NULL;
        GstElement *appsrc = NULL;
-       gchar *media = NULL;
+       const gchar *media_type = NULL;
        int av_idx;
 
-       gst_structure_get(gst_caps_get_structure(gst_pad_get_current_caps(pad), 0), "media", G_TYPE_STRING, &media, NULL);
+       gst_structure_get(gst_caps_get_structure(gst_pad_get_current_caps(pad), 0), "media", G_TYPE_STRING, &media_type, NULL);
 
-       if (!g_strrstr(media, "audio") && !g_strrstr(media, "video")) {
-               LOG_ERROR("not supported media type [%s]", media);
+       if(!_is_supported_media_type(media_type))
                return GST_PAD_PROBE_OK;
-       }
 
-       av_idx = GET_AV_IDX(g_strrstr(media, "audio"));
+       av_idx = GET_AV_IDX(_is_audio_media_type(media_type));
 
        appsrc = gst_bin_get_by_name(source->bin, _av_tbl[av_idx].appsrc_name);
-       RET_VAL_IF(appsrc == NULL, GST_PAD_PROBE_OK, "There is no appsrc for [%s]", media);
+       RET_VAL_IF(appsrc == NULL, GST_PAD_PROBE_OK, "There is no appsrc for [%s]", media_type);
 
        new_cap = gst_caps_copy(gst_pad_get_current_caps(pad));
        g_object_set(G_OBJECT(appsrc), "caps", new_cap, NULL);
 
-       LOG_INFO("setting caps for [%s appsrc] successfully", media);
+       LOG_INFO("setting caps for [%s appsrc] successfully", media_type);
        PRINT_CAPS(new_cap, "appsrc");
 
        source->filesrc_av[av_idx].sink_pad = pad;
@@ -2101,7 +2099,7 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
        int ret = WEBRTC_ERROR_NONE;
        webrtc_gst_slot_s *source = data;
        const gchar *media_type = NULL;
-       gboolean is_audio;
+       bool is_audio;
        int av_idx;
        GstElement *queue = NULL;
 
@@ -2111,14 +2109,12 @@ static void __filesrc_pipeline_decodebin_pad_added_cb(GstElement *element, GstPa
        media_type = gst_structure_get_name(gst_caps_get_structure(gst_pad_get_current_caps(pad), 0));
        RET_IF(media_type == NULL, "media_type is NULL");
 
-       LOG_INFO("[%s] new_pad[%s] media_type[%s]", GST_ELEMENT_NAME(element), GST_PAD_NAME(pad), media_type);
-
-       if (!g_strrstr(media_type, "audio") && !g_strrstr(media_type, "video")) {
-               LOG_ERROR("not supported media type [%s]", media_type);
+       if(!_is_supported_media_type(media_type))
                return;
-       }
 
-       is_audio = (g_strrstr(media_type, "audio")) ? TRUE : FALSE;
+       LOG_INFO("[%s] new_pad[%s] media_type[%s]", GST_ELEMENT_NAME(element), GST_PAD_NAME(pad), media_type);
+
+       is_audio = _is_audio_media_type(media_type);
        av_idx = GET_AV_IDX(is_audio);
 
        if (source->av[av_idx].src_pad_probe_id > 0) {
@@ -4052,7 +4048,10 @@ static void __loopback_decodebin_pad_added_cb(GstElement *decodebin, GstPad *new
        media_type = gst_structure_get_name(gst_caps_get_structure(gst_pad_get_current_caps(new_pad), 0));
        LOG_INFO("source_id[%u], media_type[%s], new_pad[%s]", source->id, media_type, GST_PAD_NAME(new_pad));
 
-       if (g_strrstr(media_type, "audio")) {
+       if(!_is_supported_media_type(media_type))
+               return;
+
+       if (_is_audio_media_type(media_type)) {
                ret = __build_loopback_audiosink(source, decodebin);
                if (ret != WEBRTC_ERROR_NONE)
                        SAFE_GST_OBJECT_UNREF(source->av[AV_IDX_AUDIO].render.pipeline);
@@ -4060,7 +4059,7 @@ static void __loopback_decodebin_pad_added_cb(GstElement *decodebin, GstPad *new
                        GENERATE_DOT(source->webrtc, source->av[AV_IDX_AUDIO].render.pipeline, "%s.%s-%s",
                                GST_ELEMENT_NAME(source->av[AV_IDX_AUDIO].render.pipeline), GST_ELEMENT_NAME(decodebin), GST_PAD_NAME(new_pad));
 
-       } else if (g_strrstr(media_type, "video")) {
+       } else {
                ret = __build_loopback_videosink(source, decodebin);
                if (ret != WEBRTC_ERROR_NONE)
                        SAFE_GST_OBJECT_UNREF(source->av[AV_IDX_VIDEO].render.pipeline);
@@ -4068,9 +4067,6 @@ static void __loopback_decodebin_pad_added_cb(GstElement *decodebin, GstPad *new
                        GENERATE_DOT(source->webrtc, source->av[AV_IDX_VIDEO].render.pipeline, "%s.%s-%s",
                                GST_ELEMENT_NAME(source->av[AV_IDX_VIDEO].render.pipeline), GST_ELEMENT_NAME(decodebin), GST_PAD_NAME(new_pad));
 
-       } else {
-               LOG_ERROR("not supported media type[%s]", media_type);
-               return;
        }
 
        if (ret != WEBRTC_ERROR_NONE) {