webrtc_sink/source: Replace MALLOC_AND_INIT_SLOT() with functions

Unnecessary variables are also removed.

[Version] 0.3.109
[Issue Type] Refactoring

Change-Id: I3a1f6f69bf89801064b6cba8b4b2bac103166e2d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index e7c030c8d9bc7aaeee363b8d03444fcdf93c88c6..a1e560fdad4a03f0449e24417d97e9ba65c9c2db 100644 (file)
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -194,14 +194,6 @@ do { \
                gst_caps_unref(x_elem_info.src_caps); \
 } while (0)
 
-#define MALLOC_AND_INIT_SLOT(x_slot, x_id, x_bin_name, x_webrtc) \
-do { \
-       x_slot = g_new0(webrtc_gst_slot_s, 1); \
-       x_slot->id = x_id; \
-       x_slot->bin = GST_BIN(gst_bin_new(x_bin_name)); \
-       x_slot->webrtc = x_webrtc; \
-} while (0)
-
 #define GENERATE_DOT(x_webrtc, x_pipeline, x_fmt, x_arg...) \
 do { \
        gchar *dot_name; \

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 5186e8538e5e0a4d1e5ddc99ea525ba202499c63..49cdc0c04ee6351e9451bdd6dbab807f1e01fdc8 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:    0.3.108
+Version:    0.3.109
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_sink.c b/src/webrtc_sink.c
index be983f5562bca7221d5999647fbf8c0a9912e0d1..11b0077b6882ae63e03d969f4a6417d8e6210167 100644 (file)
--- a/src/webrtc_sink.c
+++ b/src/webrtc_sink.c
@@ -520,10 +520,32 @@ void _sink_slot_destroy_cb(gpointer data)
        g_free(sink);
 }
 
-int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
+static int __alloc_sink_slot(webrtc_s *webrtc, const gchar *pad_name, bool forward, webrtc_gst_slot_s **sink, gchar **name)
 {
-       int ret = WEBRTC_ERROR_NONE;
        unsigned int id;
+
+       RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+       RET_VAL_IF(pad_name == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "pad_name is NULL");
+       RET_VAL_IF(sink == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
+       RET_VAL_IF(name == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "name is NULL");
+       RET_VAL_IF(webrtc->gst.sink_slots == NULL, WEBRTC_ERROR_INVALID_OPERATION, "sink_slots is NULL");
+
+       /* name/sink will be freed by function which is set to g_hash_table_new_full() */
+       id = __get_id_from_name(pad_name);
+       *name = g_strdup_printf(forward ? "forwarding_%u" : "rendering_%u", ++id);
+
+       *sink = g_new0(webrtc_gst_slot_s, 1);
+       (*sink)->id = id;
+       (*sink)->bin = GST_BIN(gst_bin_new(*name));
+       (*sink)->webrtc = webrtc;
+
+       LOG_DEBUG("webrtc[%p] sink[%p, id:%u, name:%s]", webrtc, *sink, id, *name);
+
+       return WEBRTC_ERROR_NONE;
+}
+
+int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
+{
        gchar *bin_name;
        gchar *track_name;
        webrtc_gst_slot_s *sink;
@@ -533,12 +555,10 @@ int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
        RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
        RET_VAL_IF(src_pad == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "src_pad is NULL");
 
-       /* track_name/sink will be freed by function which is set to g_hash_table_new_full() */
-       id = __get_id_from_name(GST_PAD_NAME(src_pad));
-       bin_name = g_strdup_printf("rendering_%u", ++id);
-       track_name = g_strdup_printf("track_%u", id);
+       if (__alloc_sink_slot(webrtc, GST_PAD_NAME(src_pad), false, &sink, &bin_name) != WEBRTC_ERROR_NONE)
+               return WEBRTC_ERROR_INVALID_OPERATION;
 
-       MALLOC_AND_INIT_SLOT(sink, id, bin_name, webrtc);
+       track_name = g_strdup_printf("track_%u", sink->id);
 
        g_free(bin_name);
 
@@ -554,12 +574,10 @@ int _add_rendering_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
        g_signal_connect(decodebin, "autoplug-select", G_CALLBACK(_decodebin_autoplug_select_cb), webrtc);
        g_signal_connect(decodebin, "element-added", G_CALLBACK(__decodebin_element_added_cb), sink);
 
-       ret = _add_no_target_ghostpad_to_slot(sink, false, &sink_pad);
-       if (ret != WEBRTC_ERROR_NONE)
+       if (_add_no_target_ghostpad_to_slot(sink, false, &sink_pad) != WEBRTC_ERROR_NONE)
                goto error_before_insert;
 
-       ret = _set_ghost_pad_target(sink_pad, decodebin, false);
-       if (ret != WEBRTC_ERROR_NONE)
+       if (_set_ghost_pad_target(sink_pad, decodebin, false) != WEBRTC_ERROR_NONE)
                goto error_before_insert;
 
        if (!gst_bin_add(GST_BIN(webrtc->gst.pipeline), GST_ELEMENT(sink->bin))) {
@@ -865,8 +883,6 @@ static GstCaps* __make_caps_if_h264_or_h265(GstPad *pad)
 
 int _add_forwarding_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
 {
-       int ret = WEBRTC_ERROR_NONE;
-       unsigned int id;
        gchar *bin_name;
        gchar *track_name;
        element_info_s elem_info;
@@ -880,13 +896,10 @@ int _add_forwarding_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
        RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
        RET_VAL_IF(src_pad == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "src_pad is NULL");
 
-       /* track_name/sink will be freed by function which is set to g_hash_table_new_full() */
-       id = __get_id_from_name(GST_PAD_NAME(src_pad));
-       bin_name = g_strdup_printf("forwarding_%u", ++id);
-       track_name = g_strdup_printf("track_%u", id);
-
-       MALLOC_AND_INIT_SLOT(sink, id, bin_name, webrtc);
+       if (__alloc_sink_slot(webrtc, GST_PAD_NAME(src_pad), true, &sink, &bin_name) != WEBRTC_ERROR_NONE)
+               return WEBRTC_ERROR_INVALID_OPERATION;
 
+       track_name = g_strdup_printf("track_%u", sink->id);
        g_free(bin_name);
 
        CREATE_ELEMENT_FROM_REGISTRY(elem_info, GST_KLASS_NAME_DEPAYLOADER_RTP, gst_pad_get_current_caps(src_pad), NULL, NULL, depayloader);
@@ -913,12 +926,10 @@ int _add_forwarding_sink_bin(webrtc_s *webrtc, GstPad *src_pad, bool is_audio)
 
        gst_bin_add_many(sink->bin, depayloader, capsfilter, fakesink, NULL);
 
-       ret = _add_no_target_ghostpad_to_slot(sink, false, &sink_pad);
-       if (ret != WEBRTC_ERROR_NONE)
+       if (_add_no_target_ghostpad_to_slot(sink, false, &sink_pad) != WEBRTC_ERROR_NONE)
                goto error_before_insert;
 
-       ret = _set_ghost_pad_target(sink_pad, depayloader, false);
-       if (ret != WEBRTC_ERROR_NONE)
+       if (_set_ghost_pad_target(sink_pad, depayloader, false) != WEBRTC_ERROR_NONE)
                goto error_before_insert;
 
        if (!gst_element_link_many(depayloader, capsfilter, fakesink, NULL)) {

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index 7444595184bf7f7cb84f8db77986f1e0756873c1..00f23495e61d19f554c0a54115c692f0313d1ffb 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -2622,31 +2622,45 @@ exit:
        return ret;
 }
 
-static int __add_media_source(webrtc_s *webrtc, int type, unsigned int *source_id)
+static int __alloc_source_slot(webrtc_s *webrtc, int type, webrtc_gst_slot_s **source, gchar **name)
 {
-       int ret = WEBRTC_ERROR_NONE;
        unsigned int id;
-       webrtc_gst_slot_s *source = NULL;
-       gchar *bin_name = NULL;
 
        RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
-       RET_VAL_IF(source_id == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is NULL");
+       RET_VAL_IF(source == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source is NULL");
+       RET_VAL_IF(name == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "name is NULL");
        RET_VAL_IF(webrtc->gst.source_slots == NULL, WEBRTC_ERROR_INVALID_OPERATION, "source_slots is NULL");
 
-       /* bin_name/source will be freed by function which is set to g_hash_table_new_full() */
+       /* name/source will be freed by function which is set to g_hash_table_new_full() */
        id = __get_unoccupied_id(webrtc->gst.source_slots);
        RET_VAL_IF(id == 0, WEBRTC_ERROR_INVALID_OPERATION, "source_slots are full");
 
-       bin_name = g_strdup_printf("media_source_%u", id);
+       *name = g_strdup_printf("media_source_%u", id);
 
-       MALLOC_AND_INIT_SLOT(source, id, bin_name, webrtc);
-       source->type = type;
+       *source = g_new0(webrtc_gst_slot_s, 1);
+       (*source)->id = id;
+       (*source)->bin = GST_BIN(gst_bin_new(*name));
+       (*source)->type = type;
+       (*source)->webrtc = webrtc;
 
-       ret = __build_source_bin(webrtc, source);
-       if (ret != WEBRTC_ERROR_NONE) {
-               LOG_ERROR("failed to __build_source_bin()");
+       LOG_DEBUG("webrtc[%p] source[%p, id:%u, type:%d, name:%s]", webrtc, *source, id, type, *name);
+
+       return WEBRTC_ERROR_NONE;
+}
+
+static int __add_media_source(webrtc_s *webrtc, int type, unsigned int *source_id)
+{
+       webrtc_gst_slot_s *source = NULL;
+       gchar *bin_name = NULL;
+
+       RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
+       RET_VAL_IF(source_id == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is NULL");
+
+       if (__alloc_source_slot(webrtc, type, &source, &bin_name) != WEBRTC_ERROR_NONE)
+               goto error;
+
+       if (__build_source_bin(webrtc, source) != WEBRTC_ERROR_NONE)
                goto error;
-       }
 
        if (!gst_bin_add(GST_BIN(webrtc->gst.pipeline), GST_ELEMENT(source->bin))) {
                LOG_ERROR("failed to gst_bin_add(), [%s] -> [%s] pipeline", GST_ELEMENT_NAME(source->bin), GST_ELEMENT_NAME(webrtc->gst.pipeline));
@@ -2667,7 +2681,7 @@ static int __add_media_source(webrtc_s *webrtc, int type, unsigned int *source_i
                goto error_after_insert;
        }
 
-       *source_id = id;
+       *source_id = source->id;
 
        LOG_INFO("added a source slot[%p, id:%u]", source, source->id);