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/*
* Opus Depayloader Gst Element
*
* @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* 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.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpopusdepay.h"
GST_DEBUG_CATEGORY_STATIC (rtpopusdepay_debug);
#define GST_CAT_DEFAULT (rtpopusdepay_debug)
static GstStaticPadTemplate gst_rtp_opus_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ","
"clock-rate = (int) 48000, "
"encoding-name = (string) \"X-GST-OPUS-DRAFT-SPITTKA-00\"")
);
static GstStaticPadTemplate gst_rtp_opus_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus")
);
static GstBuffer *gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
GstBuffer * buf);
static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
G_DEFINE_TYPE (GstRTPOpusDepay, gst_rtp_opus_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
static void
gst_rtp_opus_depay_class_init (GstRTPOpusDepayClass * klass)
{
GstRTPBaseDepayloadClass *gstbasertpdepayload_class;
GstElementClass *element_class;
element_class = GST_ELEMENT_CLASS (klass);
gstbasertpdepayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_opus_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_opus_depay_sink_template));
gst_element_class_set_static_metadata (element_class,
"RTP Opus packet depayloader", "Codec/Depayloader/Network/RTP",
"Extracts Opus audio from RTP packets",
"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
gstbasertpdepayload_class->process = gst_rtp_opus_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_opus_depay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpopusdepay_debug, "rtpopusdepay", 0,
"Opus RTP Depayloader");
}
static void
gst_rtp_opus_depay_init (GstRTPOpusDepay * rtpopusdepay)
{
}
static gboolean
gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
gboolean ret;
srccaps = gst_caps_new_empty_simple ("audio/x-opus");
ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
GST_DEBUG_OBJECT (depayload,
"set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
gst_caps_unref (srccaps);
depayload->clock_rate = 48000;
return ret;
}
static GstBuffer *
gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf;
GstRTPBuffer rtpbuf = { NULL, };
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf);
outbuf = gst_rtp_buffer_get_payload_buffer (&rtpbuf);
gst_rtp_buffer_unmap (&rtpbuf);
return outbuf;
}
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