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/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@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 <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpmp2tpay.h"
/* elementfactory information */
static const GstElementDetails gst_rtp_mp2t_pay_details =
GST_ELEMENT_DETAILS ("RTP MPEG2 Transport Stream payloader",
"Codec/Payloader/Network",
"Payload-encodes MPEG2 TS into RTP packets (RFC 2250)",
"Wim Taymans <wim.taymans@gmail.com>");
static GstStaticPadTemplate gst_rtp_mp2t_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/mpegts,"
"packetsize=(int)188," "systemstream=(boolean)true")
);
static GstStaticPadTemplate gst_rtp_mp2t_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"video\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"MP2T-ES\"")
);
static gboolean gst_rtp_mp2t_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_mp2t_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static GstFlowReturn gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay);
static void gst_rtp_mp2t_pay_finalize (GObject * object);
GST_BOILERPLATE (GstRTPMP2TPay, gst_rtp_mp2t_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_mp2t_pay_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_mp2t_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp2t_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_mp2t_pay_details);
}
static void
gst_rtp_mp2t_pay_class_init (GstRTPMP2TPayClass * klass)
{
GObjectClass *gobject_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gobject_class->finalize = gst_rtp_mp2t_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_mp2t_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_mp2t_pay_handle_buffer;
}
static void
gst_rtp_mp2t_pay_init (GstRTPMP2TPay * rtpmp2tpay, GstRTPMP2TPayClass * klass)
{
GST_BASE_RTP_PAYLOAD (rtpmp2tpay)->clock_rate = 90000;
GST_BASE_RTP_PAYLOAD_PT (rtpmp2tpay) = GST_RTP_PAYLOAD_MP2T;
rtpmp2tpay->adapter = gst_adapter_new ();
}
static void
gst_rtp_mp2t_pay_finalize (GObject * object)
{
GstRTPMP2TPay *rtpmp2tpay;
rtpmp2tpay = GST_RTP_MP2T_PAY (object);
g_object_unref (rtpmp2tpay->adapter);
rtpmp2tpay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_mp2t_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gst_basertppayload_set_options (payload, "video", TRUE, "MP2T-ES", 90000);
gst_basertppayload_set_outcaps (payload, NULL);
return TRUE;
}
static GstFlowReturn
gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
{
guint avail;
guint8 *payload;
GstFlowReturn ret;
GstBuffer *outbuf;
avail = gst_adapter_available (rtpmp2tpay->adapter);
outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
/* get payload */
payload = gst_rtp_buffer_get_payload (outbuf);
/* copy stuff from adapter to payload */
gst_adapter_copy (rtpmp2tpay->adapter, payload, 0, avail);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp2tpay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp2tpay), outbuf);
/* flush the adapter content */
gst_adapter_flush (rtpmp2tpay->adapter, avail);
return ret;
}
static GstFlowReturn
gst_rtp_mp2t_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRTPMP2TPay *rtpmp2tpay;
guint size, avail, packet_len;
GstClockTime timestamp, duration;
GstFlowReturn ret;
rtpmp2tpay = GST_RTP_MP2T_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
ret = GST_FLOW_OK;
avail = gst_adapter_available (rtpmp2tpay->adapter);
/* Initialize new RTP payload */
if (avail == 0) {
rtpmp2tpay->first_ts = timestamp;
rtpmp2tpay->duration = duration;
}
/* get packet length of previous data and this new data,
* payload length includes a 4 byte header */
packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtpmp2tpay->duration + duration)) {
ret = gst_rtp_mp2t_pay_flush (rtpmp2tpay);
rtpmp2tpay->first_ts = timestamp;
rtpmp2tpay->duration = duration;
/* keep filling the payload */
} else {
if (GST_CLOCK_TIME_IS_VALID (duration))
rtpmp2tpay->duration += duration;
}
/* copy buffer to adapter */
gst_adapter_push (rtpmp2tpay->adapter, buffer);
return ret;
}
gboolean
gst_rtp_mp2t_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp2tpay",
GST_RANK_NONE, GST_TYPE_RTP_MP2T_PAY);
}
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