1 /* GStreamer
2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
3 *
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.
8 *
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.
13 *
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., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
18 */
19
20 #ifdef HAVE_CONFIG_H
21 # include "config.h"
22 #endif
23
24 #include <stdlib.h>
25 #include <string.h>
26 #include <gst/rtp/gstrtpbuffer.h>
27
28 #include "gstrtpmp2tpay.h"
29 #include "gstrtputils.h"
30
31 static GstStaticPadTemplate gst_rtp_mp2t_pay_sink_template =
32 GST_STATIC_PAD_TEMPLATE ("sink",
33 GST_PAD_SINK,
34 GST_PAD_ALWAYS,
35 GST_STATIC_CAPS ("video/mpegts,"
36 "packetsize=(int)188," "systemstream=(boolean)true")
37 );
38
39 static GstStaticPadTemplate gst_rtp_mp2t_pay_src_template =
40 GST_STATIC_PAD_TEMPLATE ("src",
41 GST_PAD_SRC,
42 GST_PAD_ALWAYS,
43 GST_STATIC_CAPS ("application/x-rtp, "
44 "media = (string) \"video\", "
45 "payload = (int) " GST_RTP_PAYLOAD_MP2T_STRING ", "
46 "clock-rate = (int) 90000, " "encoding-name = (string) \"MP2T\" ; "
47 "application/x-rtp, "
48 "media = (string) \"video\", "
49 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
50 "clock-rate = (int) 90000, " "encoding-name = (string) \"MP2T\"")
51 );
52
53 static gboolean gst_rtp_mp2t_pay_setcaps (GstRTPBasePayload * payload,
54 GstCaps * caps);
55 static GstFlowReturn gst_rtp_mp2t_pay_handle_buffer (GstRTPBasePayload *
56 payload, GstBuffer * buffer);
57 static GstFlowReturn gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay);
58 static void gst_rtp_mp2t_pay_finalize (GObject * object);
59
60 #define gst_rtp_mp2t_pay_parent_class parent_class
61 G_DEFINE_TYPE (GstRTPMP2TPay, gst_rtp_mp2t_pay, GST_TYPE_RTP_BASE_PAYLOAD);
62
63 static void
gst_rtp_mp2t_pay_class_init(GstRTPMP2TPayClass * klass)64 gst_rtp_mp2t_pay_class_init (GstRTPMP2TPayClass * klass)
65 {
66 GObjectClass *gobject_class;
67 GstElementClass *gstelement_class;
68 GstRTPBasePayloadClass *gstrtpbasepayload_class;
69
70 gobject_class = (GObjectClass *) klass;
71 gstelement_class = (GstElementClass *) klass;
72 gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
73
74 gobject_class->finalize = gst_rtp_mp2t_pay_finalize;
75
76 gstrtpbasepayload_class->set_caps = gst_rtp_mp2t_pay_setcaps;
77 gstrtpbasepayload_class->handle_buffer = gst_rtp_mp2t_pay_handle_buffer;
78
79 gst_element_class_add_static_pad_template (gstelement_class,
80 &gst_rtp_mp2t_pay_sink_template);
81 gst_element_class_add_static_pad_template (gstelement_class,
82 &gst_rtp_mp2t_pay_src_template);
83 gst_element_class_set_static_metadata (gstelement_class,
84 "RTP MPEG2 Transport Stream payloader", "Codec/Payloader/Network/RTP",
85 "Payload-encodes MPEG2 TS into RTP packets (RFC 2250)",
86 "Wim Taymans <wim.taymans@gmail.com>");
87 }
88
89 static void
gst_rtp_mp2t_pay_init(GstRTPMP2TPay * rtpmp2tpay)90 gst_rtp_mp2t_pay_init (GstRTPMP2TPay * rtpmp2tpay)
91 {
92 GST_RTP_BASE_PAYLOAD (rtpmp2tpay)->clock_rate = 90000;
93 GST_RTP_BASE_PAYLOAD_PT (rtpmp2tpay) = GST_RTP_PAYLOAD_MP2T;
94
95 rtpmp2tpay->adapter = gst_adapter_new ();
96 }
97
98 static void
gst_rtp_mp2t_pay_finalize(GObject * object)99 gst_rtp_mp2t_pay_finalize (GObject * object)
100 {
101 GstRTPMP2TPay *rtpmp2tpay;
102
103 rtpmp2tpay = GST_RTP_MP2T_PAY (object);
104
105 g_object_unref (rtpmp2tpay->adapter);
106 rtpmp2tpay->adapter = NULL;
107
108 G_OBJECT_CLASS (parent_class)->finalize (object);
109 }
110
111 static gboolean
gst_rtp_mp2t_pay_setcaps(GstRTPBasePayload * payload,GstCaps * caps)112 gst_rtp_mp2t_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
113 {
114 gboolean res;
115
116 gst_rtp_base_payload_set_options (payload, "video",
117 payload->pt != GST_RTP_PAYLOAD_MP2T, "MP2T", 90000);
118 res = gst_rtp_base_payload_set_outcaps (payload, NULL);
119
120 return res;
121 }
122
123 static GstFlowReturn
gst_rtp_mp2t_pay_flush(GstRTPMP2TPay * rtpmp2tpay)124 gst_rtp_mp2t_pay_flush (GstRTPMP2TPay * rtpmp2tpay)
125 {
126 guint avail, mtu;
127 GstFlowReturn ret = GST_FLOW_OK;
128 GstBuffer *outbuf;
129
130 avail = gst_adapter_available (rtpmp2tpay->adapter);
131
132 mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp2tpay);
133
134 while (avail > 0 && (ret == GST_FLOW_OK)) {
135 guint towrite;
136 guint payload_len;
137 guint packet_len;
138 GstBuffer *paybuf;
139
140 /* this will be the total length of the packet */
141 packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
142
143 /* fill one MTU or all available bytes */
144 towrite = MIN (packet_len, mtu);
145
146 /* this is the payload length */
147 payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
148 payload_len -= payload_len % 188;
149
150 /* need whole packets */
151 if (!payload_len)
152 break;
153
154 /* create buffer to hold the payload */
155 outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
156
157 /* get payload */
158 paybuf = gst_adapter_take_buffer_fast (rtpmp2tpay->adapter, payload_len);
159 gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmp2tpay), outbuf, paybuf, 0);
160 outbuf = gst_buffer_append (outbuf, paybuf);
161 avail -= payload_len;
162
163 GST_BUFFER_PTS (outbuf) = rtpmp2tpay->first_ts;
164 GST_BUFFER_DURATION (outbuf) = rtpmp2tpay->duration;
165
166 GST_DEBUG_OBJECT (rtpmp2tpay, "pushing buffer of size %u",
167 (guint) gst_buffer_get_size (outbuf));
168
169 ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp2tpay), outbuf);
170 }
171
172 return ret;
173 }
174
175 static GstFlowReturn
gst_rtp_mp2t_pay_handle_buffer(GstRTPBasePayload * basepayload,GstBuffer * buffer)176 gst_rtp_mp2t_pay_handle_buffer (GstRTPBasePayload * basepayload,
177 GstBuffer * buffer)
178 {
179 GstRTPMP2TPay *rtpmp2tpay;
180 guint size, avail, packet_len;
181 GstClockTime timestamp, duration;
182 GstFlowReturn ret;
183
184 rtpmp2tpay = GST_RTP_MP2T_PAY (basepayload);
185
186 size = gst_buffer_get_size (buffer);
187 timestamp = GST_BUFFER_PTS (buffer);
188 duration = GST_BUFFER_DURATION (buffer);
189
190 again:
191 ret = GST_FLOW_OK;
192 avail = gst_adapter_available (rtpmp2tpay->adapter);
193
194 /* Initialize new RTP payload */
195 if (avail == 0) {
196 rtpmp2tpay->first_ts = timestamp;
197 rtpmp2tpay->duration = duration;
198 }
199
200 /* get packet length of previous data and this new data */
201 packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
202
203 /* if this buffer is going to overflow the packet, flush what we have,
204 * or if upstream is handing us several packets, to keep latency low */
205 if (!size || gst_rtp_base_payload_is_filled (basepayload,
206 packet_len, rtpmp2tpay->duration + duration)) {
207 ret = gst_rtp_mp2t_pay_flush (rtpmp2tpay);
208 rtpmp2tpay->first_ts = timestamp;
209 rtpmp2tpay->duration = duration;
210
211 /* keep filling the payload */
212 } else {
213 if (GST_CLOCK_TIME_IS_VALID (duration))
214 rtpmp2tpay->duration += duration;
215 }
216
217 /* copy buffer to adapter */
218 if (buffer) {
219 gst_adapter_push (rtpmp2tpay->adapter, buffer);
220 buffer = NULL;
221 }
222
223 if (size >= (188 * 2)) {
224 size = 0;
225 goto again;
226 }
227
228 return ret;
229
230 }
231
232 gboolean
gst_rtp_mp2t_pay_plugin_init(GstPlugin * plugin)233 gst_rtp_mp2t_pay_plugin_init (GstPlugin * plugin)
234 {
235 return gst_element_register (plugin, "rtpmp2tpay",
236 GST_RANK_SECONDARY, GST_TYPE_RTP_MP2T_PAY);
237 }
238