3 * Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.com>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
18 * Boston, MA 02110-1301, USA.
22 * SECTION:element-rtpstreampay
24 * Implements stream payloading of RTP and RTCP packets for connection-oriented
25 * transport protocols according to RFC4571.
27 * <title>Example launch line</title>
29 * gst-launch-1.0 audiotestsrc ! "audio/x-raw,rate=48000" ! vorbisenc ! rtpvorbispay config-interval=1 ! rtpstreampay ! tcpserversink port=5678
30 * gst-launch-1.0 tcpclientsrc port=5678 host=127.0.0.1 do-timestamp=true ! "application/x-rtp-stream,media=audio,clock-rate=48000,encoding-name=VORBIS" ! rtpstreamdepay ! rtpvorbisdepay ! decodebin ! audioconvert ! audioresample ! autoaudiosink
39 #include "gstrtpstreampay.h"
41 #define GST_CAT_DEFAULT gst_rtp_stream_pay_debug
42 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
44 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
47 GST_STATIC_CAPS ("application/x-rtp; application/x-rtcp; "
48 "application/x-srtp; application/x-srtcp")
51 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
54 GST_STATIC_CAPS ("application/x-rtp-stream; application/x-rtcp-stream; "
55 "application/x-srtp-stream; application/x-srtcp-stream")
58 #define parent_class gst_rtp_stream_pay_parent_class
59 G_DEFINE_TYPE (GstRtpStreamPay, gst_rtp_stream_pay, GST_TYPE_ELEMENT);
61 static gboolean gst_rtp_stream_pay_sink_query (GstPad * pad, GstObject * parent,
63 static GstFlowReturn gst_rtp_stream_pay_sink_chain (GstPad * pad,
64 GstObject * parent, GstBuffer * inbuf);
65 static gboolean gst_rtp_stream_pay_sink_event (GstPad * pad, GstObject * parent,
69 gst_rtp_stream_pay_class_init (GstRtpStreamPayClass * klass)
71 GstElementClass *gstelement_class;
73 GST_DEBUG_CATEGORY_INIT (gst_rtp_stream_pay_debug, "rtpstreampay", 0,
74 "RTP stream payloader");
76 gstelement_class = (GstElementClass *) klass;
78 gst_element_class_set_static_metadata (gstelement_class,
79 "RTP Stream Payloading", "Codec/Payloader/Network",
80 "Payloads RTP/RTCP packets for streaming protocols according to RFC4571",
81 "Sebastian Dröge <sebastian@centricular.com>");
83 gst_element_class_add_pad_template (gstelement_class,
84 gst_static_pad_template_get (&src_template));
85 gst_element_class_add_pad_template (gstelement_class,
86 gst_static_pad_template_get (&sink_template));
90 gst_rtp_stream_pay_init (GstRtpStreamPay * self)
92 self->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
93 gst_pad_set_chain_function (self->sinkpad,
94 GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_chain));
95 gst_pad_set_event_function (self->sinkpad,
96 GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_event));
97 gst_pad_set_query_function (self->sinkpad,
98 GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_query));
99 gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
101 self->srcpad = gst_pad_new_from_static_template (&src_template, "src");
102 gst_pad_use_fixed_caps (self->srcpad);
103 gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
107 gst_rtp_stream_pay_sink_get_caps (GstRtpStreamPay * self, GstCaps * filter)
109 GstCaps *peerfilter = NULL, *peercaps, *templ;
111 GstStructure *structure;
115 peerfilter = gst_caps_copy (filter);
116 n = gst_caps_get_size (peerfilter);
117 for (i = 0; i < n; i++) {
118 structure = gst_caps_get_structure (peerfilter, i);
120 if (gst_structure_has_name (structure, "application/x-rtp"))
121 gst_structure_set_name (structure, "application/x-rtp-stream");
122 else if (gst_structure_has_name (structure, "application/x-rtcp"))
123 gst_structure_set_name (structure, "application/x-rtcp-stream");
124 else if (gst_structure_has_name (structure, "application/x-srtp"))
125 gst_structure_set_name (structure, "application/x-srtp-stream");
127 gst_structure_set_name (structure, "application/x-srtcp-stream");
131 templ = gst_pad_get_pad_template_caps (self->sinkpad);
132 peercaps = gst_pad_peer_query_caps (self->srcpad, peerfilter);
135 /* Rename structure names */
136 peercaps = gst_caps_make_writable (peercaps);
137 n = gst_caps_get_size (peercaps);
138 for (i = 0; i < n; i++) {
139 structure = gst_caps_get_structure (peercaps, i);
141 if (gst_structure_has_name (structure, "application/x-rtp-stream"))
142 gst_structure_set_name (structure, "application/x-rtp");
143 else if (gst_structure_has_name (structure, "application/x-rtcp-stream"))
144 gst_structure_set_name (structure, "application/x-rtcp");
145 else if (gst_structure_has_name (structure, "application/x-srtp-stream"))
146 gst_structure_set_name (structure, "application/x-srtp");
148 gst_structure_set_name (structure, "application/x-srtcp");
151 res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
152 gst_caps_unref (peercaps);
158 GstCaps *intersection;
161 gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
162 gst_caps_unref (res);
165 gst_caps_unref (peerfilter);
172 gst_rtp_stream_pay_sink_query (GstPad * pad, GstObject * parent,
175 GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
178 GST_LOG_OBJECT (pad, "Handling query of type '%s'",
179 gst_query_type_get_name (GST_QUERY_TYPE (query)));
181 switch (GST_QUERY_TYPE (query)) {
186 gst_query_parse_caps (query, &caps);
187 caps = gst_rtp_stream_pay_sink_get_caps (self, caps);
188 gst_query_set_caps_result (query, caps);
189 gst_caps_unref (caps);
194 ret = gst_pad_query_default (pad, parent, query);
201 gst_rtp_stream_pay_sink_set_caps (GstRtpStreamPay * self, GstCaps * caps)
204 GstStructure *structure;
207 othercaps = gst_caps_copy (caps);
208 structure = gst_caps_get_structure (othercaps, 0);
210 if (gst_structure_has_name (structure, "application/x-rtp"))
211 gst_structure_set_name (structure, "application/x-rtp-stream");
212 else if (gst_structure_has_name (structure, "application/x-rtcp"))
213 gst_structure_set_name (structure, "application/x-rtcp-stream");
214 else if (gst_structure_has_name (structure, "application/x-srtp"))
215 gst_structure_set_name (structure, "application/x-srtp-stream");
217 gst_structure_set_name (structure, "application/x-srtcp-stream");
219 ret = gst_pad_set_caps (self->srcpad, othercaps);
220 gst_caps_unref (othercaps);
226 gst_rtp_stream_pay_sink_event (GstPad * pad, GstObject * parent,
229 GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
232 GST_LOG_OBJECT (pad, "Got %s event", GST_EVENT_TYPE_NAME (event));
234 switch (GST_EVENT_TYPE (event)) {
239 gst_event_parse_caps (event, &caps);
240 ret = gst_rtp_stream_pay_sink_set_caps (self, caps);
241 gst_event_unref (event);
245 ret = gst_pad_event_default (pad, parent, event);
253 gst_rtp_stream_pay_sink_chain (GstPad * pad, GstObject * parent,
256 GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
261 size = gst_buffer_get_size (inbuf);
262 if (size > G_MAXUINT16) {
263 GST_ELEMENT_ERROR (self, CORE, FAILED, (NULL),
264 ("Only buffers up to %d bytes supported, got %" G_GSIZE_FORMAT,
266 gst_buffer_unref (inbuf);
267 return GST_FLOW_ERROR;
270 outbuf = gst_buffer_new_and_alloc (2);
272 GST_WRITE_UINT16_BE (size16, size);
273 gst_buffer_fill (outbuf, 0, size16, 2);
275 gst_buffer_copy_into (outbuf, inbuf, GST_BUFFER_COPY_ALL, 0, -1);
277 gst_buffer_unref (inbuf);
279 return gst_pad_push (self->srcpad, outbuf);
283 gst_rtp_stream_pay_plugin_init (GstPlugin * plugin)
285 return gst_element_register (plugin, "rtpstreampay",
286 GST_RANK_NONE, GST_TYPE_RTP_STREAM_PAY);