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/*
* Siren Depayloader Gst Element
*
* @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpsirendepay.h"
#include "gstrtputils.h"
static GstStaticPadTemplate gst_rtp_siren_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"clock-rate = (int) 16000, " "encoding-name = (string) \"SIREN\"")
/* This is the default, so the peer doesn't have to specify it */
/* " "dct-length = (int) 320") */
);
static GstStaticPadTemplate gst_rtp_siren_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")
);
static GstBuffer *gst_rtp_siren_depay_process (GstRTPBaseDepayload *
depayload, GstRTPBuffer * rtp);
static gboolean gst_rtp_siren_depay_setcaps (GstRTPBaseDepayload *
depayload, GstCaps * caps);
G_DEFINE_TYPE (GstRTPSirenDepay, gst_rtp_siren_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
static void gst_rtp_siren_depay_class_init (GstRTPSirenDepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_siren_depay_process;
gstrtpbasedepayload_class->set_caps = gst_rtp_siren_depay_setcaps;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_siren_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_siren_depay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP Siren packet depayloader", "Codec/Depayloader/Network/RTP",
"Extracts Siren audio from RTP packets",
"Philippe Kalaf <philippe.kalaf@collabora.co.uk>");
}
static void
gst_rtp_siren_depay_init (GstRTPSirenDepay * rtpsirendepay)
{
}
static gboolean
gst_rtp_siren_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
gboolean ret;
srccaps = gst_caps_new_simple ("audio/x-siren",
"dct-length", G_TYPE_INT, 320, NULL);
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);
/* always fixed clock rate of 16000 */
depayload->clock_rate = 16000;
return ret;
}
static GstBuffer *
gst_rtp_siren_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp)
{
GstBuffer *outbuf;
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
if (outbuf) {
gst_rtp_drop_meta (GST_ELEMENT_CAST (depayload), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
}
return outbuf;
}
gboolean
gst_rtp_siren_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpsirendepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_SIREN_DEPAY);
}
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