*
* 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-rtpbvdepay
+ * @title: rtpbvdepay
+ * @see_also: rtpbvpay
+ *
+ * Extract BroadcomVoice audio from RTP packets according to RFC 4298.
+ * For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
*/
#ifdef HAVE_CONFIG_H
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/audio/audio.h>
#include "gstrtpbvdepay.h"
+#include "gstrtputils.h"
static GstStaticPadTemplate gst_rtp_bv_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"BV16\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 16000, " "encoding-name = (string) \"BV32\"")
);
GST_STATIC_CAPS ("audio/x-bv, " "mode = (int) { 16, 32 }")
);
-static GstBuffer *gst_rtp_bv_depay_process (GstBaseRTPDepayload * depayload,
- GstBuffer * buf);
-static gboolean gst_rtp_bv_depay_setcaps (GstBaseRTPDepayload * depayload,
+static GstBuffer *gst_rtp_bv_depay_process (GstRTPBaseDepayload * depayload,
+ GstRTPBuffer * rtp);
+static gboolean gst_rtp_bv_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
#define gst_rtp_bv_depay_parent_class parent_class
-G_DEFINE_TYPE (GstRTPBVDepay, gst_rtp_bv_depay, GST_TYPE_BASE_RTP_DEPAYLOAD);
+G_DEFINE_TYPE (GstRTPBVDepay, gst_rtp_bv_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
static void
gst_rtp_bv_depay_class_init (GstRTPBVDepayClass * 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_bv_depay_src_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_rtp_bv_depay_sink_template));
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_bv_depay_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_bv_depay_sink_template);
- gst_element_class_set_details_simple (gstelement_class,
+ gst_element_class_set_static_metadata (gstelement_class,
"RTP BroadcomVoice depayloader", "Codec/Depayloader/Network/RTP",
"Extracts BroadcomVoice audio from RTP packets (RFC 4298)",
"Wim Taymans <wim.taymans@collabora.co.uk>");
- gstbasertpdepayload_class->process = gst_rtp_bv_depay_process;
- gstbasertpdepayload_class->set_caps = gst_rtp_bv_depay_setcaps;
+ gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_bv_depay_process;
+ gstrtpbasedepayload_class->set_caps = gst_rtp_bv_depay_setcaps;
}
static void
}
static gboolean
-gst_rtp_bv_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+gst_rtp_bv_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstRTPBVDepay *rtpbvdepay = GST_RTP_BV_DEPAY (depayload);
GstCaps *srccaps;
srccaps = gst_caps_new_simple ("audio/x-bv",
"mode", G_TYPE_INT, rtpbvdepay->mode, NULL);
- ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+ ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
GST_DEBUG ("set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
gst_caps_unref (srccaps);
}
static GstBuffer *
-gst_rtp_bv_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+gst_rtp_bv_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstBuffer *outbuf;
gboolean marker;
- GstRTPBuffer rtp = { NULL, };
- gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+ marker = gst_rtp_buffer_get_marker (rtp);
- marker = gst_rtp_buffer_get_marker (&rtp);
+ GST_DEBUG ("process : got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
+ gst_buffer_get_size (rtp->buffer), marker,
+ gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
- GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
- gst_buffer_get_size (buf), marker,
- gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp));
-
- outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
- gst_rtp_buffer_unmap (&rtp);
+ outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
if (marker && outbuf) {
- /* mark start of talkspurt with DISCONT */
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ /* mark start of talkspurt with RESYNC */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
+ }
+
+ if (outbuf) {
+ gst_rtp_drop_non_audio_meta (depayload, outbuf);
}
return outbuf;