*
* 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_DEBUG_CATEGORY_STATIC (rtpac3depay_debug);
#define GST_CAT_DEFAULT (rtpac3depay_debug)
-/* elementfactory information */
-static const GstElementDetails gst_rtp_ac3depay_details =
-GST_ELEMENT_DETAILS ("RTP AC3 depayloader",
- "Codec/Depayloader/Network",
- "Extracts AC3 audio from RTP packets (RFC 4184)",
- "Wim Taymans <wim.taymans@gmail.com>");
-
static GstStaticPadTemplate gst_rtp_ac3_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
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\"")
);
-GST_BOILERPLATE (GstRtpAC3Depay, gst_rtp_ac3_depay, GstBaseRTPDepayload,
- 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_base_init (gpointer klass)
+gst_rtp_ac3_depay_class_init (GstRtpAC3DepayClass * klass)
{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstElementClass *gstelement_class;
+ GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
+
+ gstelement_class = (GstElementClass *) klass;
+ gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
- gst_element_class_add_pad_template (element_class,
+ 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 (element_class,
+ gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_ac3_depay_sink_template));
- gst_element_class_set_details (element_class, &gst_rtp_ac3depay_details);
-}
+ 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>");
-static void
-gst_rtp_ac3_depay_class_init (GstRtpAC3DepayClass * klass)
-{
- GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
-
- gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
-
- parent_class = g_type_class_peek_parent (klass);
-
- 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,
- "MPEG Audio RTP Depayloader");
+ "AC3 Audio RTP Depayloader");
}
static void
-gst_rtp_ac3_depay_init (GstRtpAC3Depay * rtpac3depay,
- GstRtpAC3DepayClass * klass)
+gst_rtp_ac3_depay_init (GstRtpAC3Depay * rtpac3depay)
{
- /* needed because of GST_BOILERPLATE */
+ /* needed because of G_DEFINE_TYPE */
}
static gboolean
-gst_rtp_ac3_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+gst_rtp_ac3_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
- GstRtpAC3Depay *rtpac3depay;
gint clock_rate;
GstCaps *srccaps;
gboolean res;
- rtpac3depay = GST_RTP_AC3_DEPAY (depayload);
-
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &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;
+ GstRTPBuffer rtp = { NULL, };
+ guint8 *payload;
+ guint16 FT, NF;
rtpac3depay = GST_RTP_AC3_DEPAY (depayload);
- {
- gint payload_len;
- guint8 *payload;
- guint16 FT, NF;
+ gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
- payload_len = gst_rtp_buffer_get_payload_len (buf);
- payload = gst_rtp_buffer_get_payload (buf);
+ if (gst_rtp_buffer_get_payload_len (&rtp) < 2)
+ goto empty_packet;
- if (payload_len <= 2)
- goto empty_packet;
+ payload = gst_rtp_buffer_get_payload (&rtp);
- /* strip off header
- *
- * 0 1
- * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
- * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- * | MBZ | FT| NF |
- * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- */
- FT = payload[0] & 0x3;
- NF = payload[1];
+ /* strip off header
+ *
+ * 0 1
+ * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ * | MBZ | FT| NF |
+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ */
+ FT = payload[0] & 0x3;
+ NF = payload[1];
- GST_DEBUG_OBJECT (rtpac3depay, "FT: %d, NF: %d", FT, NF);
+ GST_DEBUG_OBJECT (rtpac3depay, "FT: %d, NF: %d", FT, NF);
- payload_len -= 2;
- payload += 2;
+ /* We don't bother with fragmented packets yet */
+ outbuf = gst_rtp_buffer_get_payload_subbuffer (&rtp, 2, -1);
- /* We don't bother with fragmented packets yet */
- outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 2, -1);
+ gst_rtp_buffer_unmap (&rtp);
- GST_DEBUG_OBJECT (rtpac3depay, "pushing buffer of size %d",
- GST_BUFFER_SIZE (outbuf));
-
- return outbuf;
- }
+ if (outbuf)
+ GST_DEBUG_OBJECT (rtpac3depay, "pushing buffer of size %" G_GSIZE_FORMAT,
+ gst_buffer_get_size (outbuf));
- return NULL;
+ return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpac3depay, STREAM, DECODE,
("Empty Payload."), (NULL));
+ gst_rtp_buffer_unmap (&rtp);
return NULL;
}
}
gst_rtp_ac3_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpac3depay",
- GST_RANK_MARGINAL, GST_TYPE_RTP_AC3_DEPAY);
+ GST_RANK_SECONDARY, GST_TYPE_RTP_AC3_DEPAY);
}
projects / platform / upstream / gstreamer.git / blobdiff