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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2005> Zeeshan Ali <zeenix@gmail.com>
*
* 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 <gst/rtp/gstrtpbuffer.h>
#include "gstrtpgsmdepay.h"
GST_DEBUG_CATEGORY_STATIC (rtpgsmdepay_debug);
#define GST_CAT_DEFAULT (rtpgsmdepay_debug)
/* elementfactory information */
static GstElementDetails gst_rtp_gsmdepay_details = {
"RTP GSM depayloader",
"Codec/Depayloader/Network",
"Extracts GSM audio from RTP packets",
"Zeeshan Ali <zeenix@gmail.com>"
};
/* RTPGSMDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
static GstStaticPadTemplate gst_rtp_gsm_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = 1")
);
static GstStaticPadTemplate gst_rtp_gsm_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) 8000, " "encoding-name = (string) \"GSM\";"
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_GSM_STRING ", "
"clock-rate = (int) 8000")
);
static GstBuffer *gst_rtp_gsm_depay_process (GstBaseRTPDepayload * _depayload,
GstBuffer * buf);
static gboolean gst_rtp_gsm_depay_setcaps (GstBaseRTPDepayload * _depayload,
GstCaps * caps);
GST_BOILERPLATE (GstRTPGSMDepay, gst_rtp_gsm_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static void
gst_rtp_gsm_depay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_gsm_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_gsm_depay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_gsmdepay_details);
}
static void
gst_rtp_gsm_depay_class_init (GstRTPGSMDepayClass * klass)
{
GstBaseRTPDepayloadClass *gstbasertp_depayload_class;
gstbasertp_depayload_class = (GstBaseRTPDepayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gstbasertp_depayload_class->process = gst_rtp_gsm_depay_process;
gstbasertp_depayload_class->set_caps = gst_rtp_gsm_depay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpgsmdepay_debug, "rtpgsmdepay", 0,
"GSM Audio RTP Depayloader");
}
static void
gst_rtp_gsm_depay_init (GstRTPGSMDepay * rtpgsmdepay,
GstRTPGSMDepayClass * klass)
{
/* needed because of GST_BOILERPLATE */
}
static gboolean
gst_rtp_gsm_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
gboolean ret;
GstStructure *structure;
gint clock_rate;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = 8000; /* default */
depayload->clock_rate = clock_rate;
srccaps = gst_caps_new_simple ("audio/x-gsm",
"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, clock_rate, NULL);
ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
return ret;
}
static GstBuffer *
gst_rtp_gsm_depay_process (GstBaseRTPDepayload * _depayload, GstBuffer * buf)
{
GstBuffer *outbuf = NULL;
gboolean marker;
marker = gst_rtp_buffer_get_marker (buf);
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
GST_BUFFER_SIZE (buf), marker,
gst_rtp_buffer_get_timestamp (buf), gst_rtp_buffer_get_seq (buf));
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
if (marker) {
/* mark start of talkspurt with DISCONT */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
return outbuf;
}
gboolean
gst_rtp_gsm_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpgsmdepay",
GST_RANK_MARGINAL, GST_TYPE_RTP_GSM_DEPAY);
}
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