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206 lines
6 KiB
C
206 lines
6 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpmp2tpay.h"
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static GstStaticPadTemplate gst_rtp_mp2t_pay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("video/mpegts,"
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"packetsize=(int)188," "systemstream=(boolean)true")
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);
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static GstStaticPadTemplate gst_rtp_mp2t_pay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"video\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 90000, " "encoding-name = (string) \"MP2T-ES\"")
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);
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static gboolean gst_rtp_mp2t_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_mp2t_pay_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buffer);
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static GstFlowReturn gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay);
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static void gst_rtp_mp2t_pay_finalize (GObject * object);
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GST_BOILERPLATE (GstRTPMP2TPay, gst_rtp_mp2t_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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static void
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gst_rtp_mp2t_pay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_mp2t_pay_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_mp2t_pay_src_template));
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gst_element_class_set_details_simple (element_class,
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"RTP MPEG2 Transport Stream payloader", "Codec/Payloader/Network/RTP",
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"Payload-encodes MPEG2 TS into RTP packets (RFC 2250)",
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"Wim Taymans <wim.taymans@gmail.com>");
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}
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static void
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gst_rtp_mp2t_pay_class_init (GstRTPMP2TPayClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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gobject_class->finalize = gst_rtp_mp2t_pay_finalize;
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gstbasertppayload_class->set_caps = gst_rtp_mp2t_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_mp2t_pay_handle_buffer;
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}
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static void
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gst_rtp_mp2t_pay_init (GstRTPMP2TPay * rtpmp2tpay, GstRTPMP2TPayClass * klass)
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{
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GST_BASE_RTP_PAYLOAD (rtpmp2tpay)->clock_rate = 90000;
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GST_BASE_RTP_PAYLOAD_PT (rtpmp2tpay) = GST_RTP_PAYLOAD_MP2T;
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rtpmp2tpay->adapter = gst_adapter_new ();
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}
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static void
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gst_rtp_mp2t_pay_finalize (GObject * object)
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{
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GstRTPMP2TPay *rtpmp2tpay;
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rtpmp2tpay = GST_RTP_MP2T_PAY (object);
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g_object_unref (rtpmp2tpay->adapter);
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rtpmp2tpay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_rtp_mp2t_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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gboolean res;
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gst_basertppayload_set_options (payload, "video", TRUE, "MP2T-ES", 90000);
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res = gst_basertppayload_set_outcaps (payload, NULL);
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return res;
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}
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static GstFlowReturn
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gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
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{
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guint avail;
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guint8 *payload;
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GstFlowReturn ret;
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GstBuffer *outbuf;
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avail = gst_adapter_available (rtpmp2tpay->adapter);
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outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
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/* get payload */
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payload = gst_rtp_buffer_get_payload (outbuf);
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/* copy stuff from adapter to payload */
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gst_adapter_copy (rtpmp2tpay->adapter, payload, 0, avail);
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GST_BUFFER_TIMESTAMP (outbuf) = rtpmp2tpay->first_ts;
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GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
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GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %d",
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GST_BUFFER_SIZE (outbuf));
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp2tpay), outbuf);
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/* flush the adapter content */
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gst_adapter_flush (rtpmp2tpay->adapter, avail);
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return ret;
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}
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static GstFlowReturn
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gst_rtp_mp2t_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRTPMP2TPay *rtpmp2tpay;
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guint size, avail, packet_len;
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GstClockTime timestamp, duration;
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GstFlowReturn ret;
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rtpmp2tpay = GST_RTP_MP2T_PAY (basepayload);
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size = GST_BUFFER_SIZE (buffer);
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timestamp = GST_BUFFER_TIMESTAMP (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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ret = GST_FLOW_OK;
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avail = gst_adapter_available (rtpmp2tpay->adapter);
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/* Initialize new RTP payload */
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if (avail == 0) {
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rtpmp2tpay->first_ts = timestamp;
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rtpmp2tpay->duration = duration;
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}
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/* get packet length of previous data and this new data,
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* payload length includes a 4 byte header */
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packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0);
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/* if this buffer is going to overflow the packet, flush what we
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* have. */
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if (gst_basertppayload_is_filled (basepayload,
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packet_len, rtpmp2tpay->duration + duration)) {
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ret = gst_rtp_mp2t_pay_flush (rtpmp2tpay);
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rtpmp2tpay->first_ts = timestamp;
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rtpmp2tpay->duration = duration;
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/* keep filling the payload */
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} else {
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if (GST_CLOCK_TIME_IS_VALID (duration))
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rtpmp2tpay->duration += duration;
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}
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/* copy buffer to adapter */
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gst_adapter_push (rtpmp2tpay->adapter, buffer);
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return ret;
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}
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gboolean
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gst_rtp_mp2t_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpmp2tpay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_MP2T_PAY);
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}
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