GstCaps *sink_caps;
webrtc_gst_slot_s *sink;
GstPad *sink_pad = NULL;
- GstCaps *src_caps;
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");
g_free(bin_name);
- src_caps = gst_pad_get_current_caps(src_pad);
- CREATE_ELEMENT_FROM_REGISTRY(elem_info, GST_KLASS_NAME_DEPAYLOADER_RTP, src_caps, NULL, NULL, depayloader);
- gst_caps_unref(src_caps);
+ CREATE_ELEMENT_FROM_REGISTRY(elem_info, GST_KLASS_NAME_DEPAYLOADER_RTP, gst_pad_get_current_caps(src_pad), NULL, NULL, depayloader);
if (!depayloader)
goto error_before_insert;
static GstElement * __create_payloader_for_filesrc_pipeline(GstPad *pad, bool is_audio)
{
element_info_s elem_info;
- GstCaps *caps;
GstElement *payloader = NULL;
RET_VAL_IF(pad == NULL, NULL, "pad is NULL");
- caps = gst_pad_get_current_caps(pad);
-
CREATE_ELEMENT_FROM_REGISTRY(elem_info, GST_KLASS_NAME_PAYLOADER_RTP,
- caps,
+ gst_pad_get_current_caps(pad),
NULL,
NULL,
payloader);
-
- gst_caps_unref(caps);
RET_VAL_IF(payloader == NULL, NULL, "payloader is NULL");
gst_element_set_name(payloader, _av_tbl[GET_AV_IDX(is_audio)].payloader_name);