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/*
* Siren Payloader 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtpsirenpay.h"
#include <gst/rtp/gstrtpbuffer.h>
/* elementfactory information */
static GstElementDetails gst_rtpsirenpay_details = {
"RTP Payloader for Siren Audio",
"Codec/Payloader/Network",
"Packetize Siren audio streams into RTP packets",
"Youness Alaoui <kakaroto@kakaroto.homelinux.net>"
};
GST_DEBUG_CATEGORY_STATIC (rtpsirenpay_debug);
#define GST_CAT_DEFAULT (rtpsirenpay_debug)
static GstStaticPadTemplate gst_rtpsirenpay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")
);
static GstStaticPadTemplate gst_rtpsirenpay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 16000, "
"encoding-name = (string) \"SIREN\", "
"bitrate = (string) \"16000\", " "dct-length = (int) 320")
);
static gboolean gst_rtpsirenpay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
GST_BOILERPLATE (GstRTPSirenPay, gst_rtpsirenpay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtpsirenpay_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_rtpsirenpay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpsirenpay_src_template));
gst_element_class_set_details (element_class, &gst_rtpsirenpay_details);
}
static void
gst_rtpsirenpay_class_init (GstRTPSirenPayClass * klass)
{
GstBaseRTPPayloadClass *gstbasertppayload_class;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
gstbasertppayload_class->set_caps = gst_rtpsirenpay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpsirenpay_debug, "rtpsirenpay", 0,
"siren audio RTP payloader");
}
static void
gst_rtpsirenpay_init (GstRTPSirenPay * rtpsirenpay, GstRTPSirenPayClass * klass)
{
GstBaseRTPPayload *basertppayload;
GstBaseRTPAudioPayload *basertpaudiopayload;
basertppayload = GST_BASE_RTP_PAYLOAD (rtpsirenpay);
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpsirenpay);
/* we don't set the payload type, it should be set by the application using
* the pt property or the default 96 will be used */
basertppayload->clock_rate = 16000;
/* tell basertpaudiopayload that this is a frame based codec */
gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
}
static gboolean
gst_rtpsirenpay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
{
GstRTPSirenPay *rtpsirenpay;
GstBaseRTPAudioPayload *basertpaudiopayload;
gint dct_length;
GstStructure *structure;
const char *payload_name;
rtpsirenpay = GST_RTP_SIREN_PAY (basertppayload);
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "dct-length", &dct_length);
if (dct_length != 320)
goto wrong_dct;
payload_name = gst_structure_get_name (structure);
if (g_ascii_strcasecmp ("audio/x-siren", payload_name))
goto wrong_caps;
gst_basertppayload_set_options (basertppayload, "audio", TRUE, "SIREN",
16000);
/* set options for this frame based audio codec */
gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 20, 40);
return gst_basertppayload_set_outcaps (basertppayload, NULL);
/* ERRORS */
wrong_dct:
{
GST_ERROR_OBJECT (rtpsirenpay, "dct-length must be 320, received %d",
dct_length);
return FALSE;
}
wrong_caps:
{
GST_ERROR_OBJECT (rtpsirenpay, "expected audio/x-siren, received %s",
payload_name);
return FALSE;
}
}
gboolean
gst_rtp_siren_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpsirenpay",
GST_RANK_NONE, GST_TYPE_RTP_SIREN_PAY);
}
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