/* 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" 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_simple (element_class, "RTP MPEG2 Transport Stream payloader", "Codec/Payloader/Network", "Payload-encodes MPEG2 TS into RTP packets (RFC 2250)", "Wim Taymans <wim.taymans@gmail.com>"); } 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) { gboolean res; gst_basertppayload_set_options (payload, "video", TRUE, "MP2T-ES", 90000); res = gst_basertppayload_set_outcaps (payload, NULL); return res; } 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); }