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
23 * @title: rtpstreampay
25 * Implements stream payloading of RTP and RTCP packets for connection-oriented
26 * transport protocols according to RFC4571.
28 * ## Example launch line
30 * gst-launch-1.0 audiotestsrc ! "audio/x-raw,rate=48000" ! vorbisenc ! rtpvorbispay config-interval=1 ! rtpstreampay ! tcpserversink port=5678
31 * 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
40 #include "gstrtpstreampay.h"
42 #define GST_CAT_DEFAULT gst_rtp_stream_pay_debug
43 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
45 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
48 GST_STATIC_CAPS ("application/x-rtp; application/x-rtcp; "
49 "application/x-srtp; application/x-srtcp")
52 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
55 GST_STATIC_CAPS ("application/x-rtp-stream; application/x-rtcp-stream; "
56 "application/x-srtp-stream; application/x-srtcp-stream")
59 #define parent_class gst_rtp_stream_pay_parent_class
60 G_DEFINE_TYPE (GstRtpStreamPay, gst_rtp_stream_pay, GST_TYPE_ELEMENT);
62 static gboolean gst_rtp_stream_pay_sink_query (GstPad * pad, GstObject * parent,
64 static GstFlowReturn gst_rtp_stream_pay_sink_chain (GstPad * pad,
65 GstObject * parent, GstBuffer * inbuf);
66 static gboolean gst_rtp_stream_pay_sink_event (GstPad * pad, GstObject * parent,
70 gst_rtp_stream_pay_class_init (GstRtpStreamPayClass * klass)
72 GstElementClass *gstelement_class;
74 GST_DEBUG_CATEGORY_INIT (gst_rtp_stream_pay_debug, "rtpstreampay", 0,
75 "RTP stream payloader");
77 gstelement_class = (GstElementClass *) klass;
79 gst_element_class_set_static_metadata (gstelement_class,
80 "RTP Stream Payloading", "Codec/Payloader/Network",
81 "Payloads RTP/RTCP packets for streaming protocols according to RFC4571",
82 "Sebastian Dröge <sebastian@centricular.com>");
84 gst_element_class_add_static_pad_template (gstelement_class, &src_template);
85 gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
89 gst_rtp_stream_pay_init (GstRtpStreamPay * self)
91 self->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
92 gst_pad_set_chain_function (self->sinkpad,
93 GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_chain));
94 gst_pad_set_event_function (self->sinkpad,
95 GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_event));
96 gst_pad_set_query_function (self->sinkpad,
97 GST_DEBUG_FUNCPTR (gst_rtp_stream_pay_sink_query));
98 gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
100 self->srcpad = gst_pad_new_from_static_template (&src_template, "src");
101 gst_pad_use_fixed_caps (self->srcpad);
102 gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
106 gst_rtp_stream_pay_sink_get_caps (GstRtpStreamPay * self, GstCaps * filter)
108 GstCaps *peerfilter = NULL, *peercaps, *templ;
110 GstStructure *structure;
114 peerfilter = gst_caps_copy (filter);
115 n = gst_caps_get_size (peerfilter);
116 for (i = 0; i < n; i++) {
117 structure = gst_caps_get_structure (peerfilter, i);
119 if (gst_structure_has_name (structure, "application/x-rtp"))
120 gst_structure_set_name (structure, "application/x-rtp-stream");
121 else if (gst_structure_has_name (structure, "application/x-rtcp"))
122 gst_structure_set_name (structure, "application/x-rtcp-stream");
123 else if (gst_structure_has_name (structure, "application/x-srtp"))
124 gst_structure_set_name (structure, "application/x-srtp-stream");
126 gst_structure_set_name (structure, "application/x-srtcp-stream");
130 templ = gst_pad_get_pad_template_caps (self->sinkpad);
131 peercaps = gst_pad_peer_query_caps (self->srcpad, peerfilter);
134 /* Rename structure names */
135 peercaps = gst_caps_make_writable (peercaps);
136 n = gst_caps_get_size (peercaps);
137 for (i = 0; i < n; i++) {
138 structure = gst_caps_get_structure (peercaps, i);
140 if (gst_structure_has_name (structure, "application/x-rtp-stream"))
141 gst_structure_set_name (structure, "application/x-rtp");
142 else if (gst_structure_has_name (structure, "application/x-rtcp-stream"))
143 gst_structure_set_name (structure, "application/x-rtcp");
144 else if (gst_structure_has_name (structure, "application/x-srtp-stream"))
145 gst_structure_set_name (structure, "application/x-srtp");
147 gst_structure_set_name (structure, "application/x-srtcp");
150 res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
151 gst_caps_unref (peercaps);
157 GstCaps *intersection;
160 gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
161 gst_caps_unref (res);
164 gst_caps_unref (peerfilter);
171 gst_rtp_stream_pay_sink_query (GstPad * pad, GstObject * parent,
174 GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
177 GST_LOG_OBJECT (pad, "Handling query of type '%s'",
178 gst_query_type_get_name (GST_QUERY_TYPE (query)));
180 switch (GST_QUERY_TYPE (query)) {
185 gst_query_parse_caps (query, &caps);
186 caps = gst_rtp_stream_pay_sink_get_caps (self, caps);
187 gst_query_set_caps_result (query, caps);
188 gst_caps_unref (caps);
193 ret = gst_pad_query_default (pad, parent, query);
200 gst_rtp_stream_pay_sink_set_caps (GstRtpStreamPay * self, GstCaps * caps)
203 GstStructure *structure;
206 othercaps = gst_caps_copy (caps);
207 structure = gst_caps_get_structure (othercaps, 0);
209 if (gst_structure_has_name (structure, "application/x-rtp"))
210 gst_structure_set_name (structure, "application/x-rtp-stream");
211 else if (gst_structure_has_name (structure, "application/x-rtcp"))
212 gst_structure_set_name (structure, "application/x-rtcp-stream");
213 else if (gst_structure_has_name (structure, "application/x-srtp"))
214 gst_structure_set_name (structure, "application/x-srtp-stream");
216 gst_structure_set_name (structure, "application/x-srtcp-stream");
218 ret = gst_pad_set_caps (self->srcpad, othercaps);
219 gst_caps_unref (othercaps);
225 gst_rtp_stream_pay_sink_event (GstPad * pad, GstObject * parent,
228 GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
231 GST_LOG_OBJECT (pad, "Got %s event", GST_EVENT_TYPE_NAME (event));
233 switch (GST_EVENT_TYPE (event)) {
238 gst_event_parse_caps (event, &caps);
239 ret = gst_rtp_stream_pay_sink_set_caps (self, caps);
240 gst_event_unref (event);
244 ret = gst_pad_event_default (pad, parent, event);
252 gst_rtp_stream_pay_sink_chain (GstPad * pad, GstObject * parent,
255 GstRtpStreamPay *self = GST_RTP_STREAM_PAY (parent);
260 size = gst_buffer_get_size (inbuf);
261 if (size > G_MAXUINT16) {
262 GST_ELEMENT_ERROR (self, CORE, FAILED, (NULL),
263 ("Only buffers up to %d bytes supported, got %" G_GSIZE_FORMAT,
265 gst_buffer_unref (inbuf);
266 return GST_FLOW_ERROR;
269 outbuf = gst_buffer_new_and_alloc (2);
271 GST_WRITE_UINT16_BE (size16, size);
272 gst_buffer_fill (outbuf, 0, size16, 2);
274 gst_buffer_copy_into (outbuf, inbuf, GST_BUFFER_COPY_ALL, 0, -1);
276 gst_buffer_unref (inbuf);
278 return gst_pad_push (self->srcpad, outbuf);
282 gst_rtp_stream_pay_plugin_init (GstPlugin * plugin)
284 return gst_element_register (plugin, "rtpstreampay",
285 GST_RANK_NONE, GST_TYPE_RTP_STREAM_PAY);