* Boston, MA 02110-1301, USA.
*/
+/**
+ * SECTION:element-rtpL16pay
+ * @title: rtpL16pay
+ * @see_also: rtpL16depay
+ *
+ * Payload raw audio into RTP packets according to RFC 3551.
+ * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
+ *
+ * ## Example pipeline
+ * |[
+ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink
+ * ]| This example pipeline will payload raw audio. Refer to
+ * the rtpL16depay example to depayload and play the RTP stream.
+ *
+ */
+
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
gstrtpbasepayload_class->get_caps = gst_rtp_L16_pay_getcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer;
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_rtp_L16_pay_src_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_rtp_L16_pay_sink_template));
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_L16_pay_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_L16_pay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP audio payloader", "Codec/Payloader/Network/RTP",
if (gst_structure_get_int (structure, "channels", &channels)) {
gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
} else if (gst_structure_get_int (structure, "payload", &pt)) {
- if (pt == 10)
+ if (pt == GST_RTP_PAYLOAD_L16_STEREO)
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
- else if (pt == 11)
+ else if (pt == GST_RTP_PAYLOAD_L16_MONO)
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
}
if (gst_structure_get_int (structure, "clock-rate", &rate)) {
gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
} else if (gst_structure_get_int (structure, "payload", &pt)) {
- if (pt == 10 || pt == 11)
+ if (pt == GST_RTP_PAYLOAD_L16_STEREO || pt == GST_RTP_PAYLOAD_L16_MONO)
gst_caps_set_simple (caps, "rate", G_TYPE_INT, 44100, NULL);
}