2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
26 #include <gst/rtp/gstrtpbuffer.h>
28 #include "gstrtpmp2tpay.h"
30 static GstStaticPadTemplate gst_rtp_mp2t_pay_sink_template =
31 GST_STATIC_PAD_TEMPLATE ("sink",
34 GST_STATIC_CAPS ("video/mpegts,"
35 "packetsize=(int)188," "systemstream=(boolean)true")
38 static GstStaticPadTemplate gst_rtp_mp2t_pay_src_template =
39 GST_STATIC_PAD_TEMPLATE ("src",
42 GST_STATIC_CAPS ("application/x-rtp, "
43 "media = (string) \"video\", "
44 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
45 "clock-rate = (int) 90000, " "encoding-name = (string) \"MP2T\"")
48 static gboolean gst_rtp_mp2t_pay_setcaps (GstRTPBasePayload * payload,
50 static GstFlowReturn gst_rtp_mp2t_pay_handle_buffer (GstRTPBasePayload *
51 payload, GstBuffer * buffer);
52 static GstFlowReturn gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay);
53 static void gst_rtp_mp2t_pay_finalize (GObject * object);
55 #define gst_rtp_mp2t_pay_parent_class parent_class
56 G_DEFINE_TYPE (GstRTPMP2TPay, gst_rtp_mp2t_pay, GST_TYPE_RTP_BASE_PAYLOAD);
59 gst_rtp_mp2t_pay_class_init (GstRTPMP2TPayClass * klass)
61 GObjectClass *gobject_class;
62 GstElementClass *gstelement_class;
63 GstRTPBasePayloadClass *gstrtpbasepayload_class;
65 gobject_class = (GObjectClass *) klass;
66 gstelement_class = (GstElementClass *) klass;
67 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
69 gobject_class->finalize = gst_rtp_mp2t_pay_finalize;
71 gstrtpbasepayload_class->set_caps = gst_rtp_mp2t_pay_setcaps;
72 gstrtpbasepayload_class->handle_buffer = gst_rtp_mp2t_pay_handle_buffer;
74 gst_element_class_add_pad_template (gstelement_class,
75 gst_static_pad_template_get (&gst_rtp_mp2t_pay_sink_template));
76 gst_element_class_add_pad_template (gstelement_class,
77 gst_static_pad_template_get (&gst_rtp_mp2t_pay_src_template));
78 gst_element_class_set_details_simple (gstelement_class,
79 "RTP MPEG2 Transport Stream payloader", "Codec/Payloader/Network/RTP",
80 "Payload-encodes MPEG2 TS into RTP packets (RFC 2250)",
81 "Wim Taymans <wim.taymans@gmail.com>");
85 gst_rtp_mp2t_pay_init (GstRTPMP2TPay * rtpmp2tpay)
87 GST_RTP_BASE_PAYLOAD (rtpmp2tpay)->clock_rate = 90000;
88 GST_RTP_BASE_PAYLOAD_PT (rtpmp2tpay) = GST_RTP_PAYLOAD_MP2T;
90 rtpmp2tpay->adapter = gst_adapter_new ();
94 gst_rtp_mp2t_pay_finalize (GObject * object)
96 GstRTPMP2TPay *rtpmp2tpay;
98 rtpmp2tpay = GST_RTP_MP2T_PAY (object);
100 g_object_unref (rtpmp2tpay->adapter);
101 rtpmp2tpay->adapter = NULL;
103 G_OBJECT_CLASS (parent_class)->finalize (object);
107 gst_rtp_mp2t_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
111 gst_rtp_base_payload_set_options (payload, "video", TRUE, "MP2T", 90000);
112 res = gst_rtp_base_payload_set_outcaps (payload, NULL);
118 gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
124 GstRTPBuffer rtp = { NULL };
126 avail = gst_adapter_available (rtpmp2tpay->adapter);
129 outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
132 gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
133 payload = gst_rtp_buffer_get_payload (&rtp);
135 /* copy stuff from adapter to payload */
136 gst_adapter_copy (rtpmp2tpay->adapter, payload, 0, avail);
137 gst_rtp_buffer_unmap (&rtp);
139 GST_BUFFER_TIMESTAMP (outbuf) = rtpmp2tpay->first_ts;
140 GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
142 GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %" G_GSIZE_FORMAT,
143 gst_buffer_get_size (outbuf));
145 ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp2tpay), outbuf);
147 /* flush the adapter content */
148 gst_adapter_flush (rtpmp2tpay->adapter, avail);
154 gst_rtp_mp2t_pay_handle_buffer (GstRTPBasePayload * basepayload,
157 GstRTPMP2TPay *rtpmp2tpay;
158 guint size, avail, packet_len;
159 GstClockTime timestamp, duration;
162 rtpmp2tpay = GST_RTP_MP2T_PAY (basepayload);
164 size = gst_buffer_get_size (buffer);
165 timestamp = GST_BUFFER_TIMESTAMP (buffer);
166 duration = GST_BUFFER_DURATION (buffer);
169 avail = gst_adapter_available (rtpmp2tpay->adapter);
171 /* Initialize new RTP payload */
173 rtpmp2tpay->first_ts = timestamp;
174 rtpmp2tpay->duration = duration;
177 /* get packet length of previous data and this new data,
178 * payload length includes a 4 byte header */
179 packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0);
181 /* if this buffer is going to overflow the packet, flush what we
183 if (gst_rtp_base_payload_is_filled (basepayload,
184 packet_len, rtpmp2tpay->duration + duration)) {
185 ret = gst_rtp_mp2t_pay_flush (rtpmp2tpay);
186 rtpmp2tpay->first_ts = timestamp;
187 rtpmp2tpay->duration = duration;
189 /* keep filling the payload */
191 if (GST_CLOCK_TIME_IS_VALID (duration))
192 rtpmp2tpay->duration += duration;
195 /* copy buffer to adapter */
196 gst_adapter_push (rtpmp2tpay->adapter, buffer);
203 gst_rtp_mp2t_pay_plugin_init (GstPlugin * plugin)
205 return gst_element_register (plugin, "rtpmp2tpay",
206 GST_RANK_SECONDARY, GST_TYPE_RTP_MP2T_PAY);