*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:element-rtpac3depay
+ * @see_also: rtpac3pay
+ *
+ * Extract AC3 audio from RTP packets according to RFC 4184.
+ * For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt
+ *
+ * <refsect2>
+ * <title>Example pipeline</title>
+ * |[
+ * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)AC3, payload=(int)96' ! rtpac3depay ! a52dec ! pulsesink
+ * ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to
+ * the rtpac3pay example to create the RTP stream.
+ * </refsect2>
+ *
+ * Last reviewed on 2013-04-25 (1.1.0)
*/
#ifdef HAVE_CONFIG_H
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) { 32000, 44100, 48000 }, "
"encoding-name = (string) \"AC3\"")
);
-G_DEFINE_TYPE (GstRtpAC3Depay, gst_rtp_ac3_depay, GST_TYPE_BASE_RTP_DEPAYLOAD);
+G_DEFINE_TYPE (GstRtpAC3Depay, gst_rtp_ac3_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
-static gboolean gst_rtp_ac3_depay_setcaps (GstBaseRTPDepayload * depayload,
+static gboolean gst_rtp_ac3_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
-static GstBuffer *gst_rtp_ac3_depay_process (GstBaseRTPDepayload * depayload,
+static GstBuffer *gst_rtp_ac3_depay_process (GstRTPBaseDepayload * depayload,
GstBuffer * buf);
static void
gst_rtp_ac3_depay_class_init (GstRtpAC3DepayClass * klass)
{
GstElementClass *gstelement_class;
- GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+ GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gstelement_class = (GstElementClass *) klass;
- gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+ gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_ac3_depay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_ac3_depay_sink_template));
- gst_element_class_set_details_simple (gstelement_class, "RTP AC3 depayloader",
- "Codec/Depayloader/Network/RTP",
+ gst_element_class_set_static_metadata (gstelement_class,
+ "RTP AC3 depayloader", "Codec/Depayloader/Network/RTP",
"Extracts AC3 audio from RTP packets (RFC 4184)",
"Wim Taymans <wim.taymans@gmail.com>");
- gstbasertpdepayload_class->set_caps = gst_rtp_ac3_depay_setcaps;
- gstbasertpdepayload_class->process = gst_rtp_ac3_depay_process;
+ gstrtpbasedepayload_class->set_caps = gst_rtp_ac3_depay_setcaps;
+ gstrtpbasedepayload_class->process = gst_rtp_ac3_depay_process;
GST_DEBUG_CATEGORY_INIT (rtpac3depay_debug, "rtpac3depay", 0,
"AC3 Audio RTP Depayloader");
}
static gboolean
-gst_rtp_ac3_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+gst_rtp_ac3_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
gint clock_rate;
clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
- srccaps = gst_caps_new_simple ("audio/ac3", NULL);
+ srccaps = gst_caps_new_empty_simple ("audio/ac3");
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
};
static GstBuffer *
-gst_rtp_ac3_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+gst_rtp_ac3_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpAC3Depay *rtpac3depay;
GstBuffer *outbuf;
gst_rtp_buffer_unmap (&rtp);
if (outbuf)
- GST_DEBUG_OBJECT (rtpac3depay, "pushing buffer of size %d",
+ GST_DEBUG_OBJECT (rtpac3depay, "pushing buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (outbuf));
return outbuf;