/**
* SECTION:element-rtpbvdepay
+ * @title: rtpbvdepay
* @see_also: rtpbvpay
*
* Extract BroadcomVoice audio from RTP packets according to RFC 4298.
#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",
);
static GstBuffer *gst_rtp_bv_depay_process (GstRTPBaseDepayload * depayload,
- GstBuffer * buf);
+ GstRTPBuffer * rtp);
static gboolean gst_rtp_bv_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
gstelement_class = (GstElementClass *) 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_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>");
- gstrtpbasedepayload_class->process = gst_rtp_bv_depay_process;
+ gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_bv_depay_process;
gstrtpbasedepayload_class->set_caps = gst_rtp_bv_depay_setcaps;
}
}
static GstBuffer *
-gst_rtp_bv_depay_process (GstRTPBaseDepayload * 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 (buf), marker,
- gst_rtp_buffer_get_timestamp (&rtp), gst_rtp_buffer_get_seq (&rtp));
+ gst_buffer_get_size (rtp->buffer), 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 RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
+ if (outbuf) {
+ gst_rtp_drop_non_audio_meta (depayload, outbuf);
+ }
+
return outbuf;
}