summaryrefslogtreecommitdiff
path: root/gst/rtp/gstrtpstreamdepay.c
blob: bcaec19ea20f5577cfb05ffd856c97982ff32066 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
/* GStreamer
 * Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.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., 51 Franklin St, Fifth Floor,
 * Boston, MA 02110-1301, USA.
 */

/**
 * SECTION:element-rtpstreamdepay
 * @title: rtpstreamdepay
 *
 * Implements stream depayloading of RTP and RTCP packets for connection-oriented
 * transport protocols according to RFC4571.
 *
 * ## Example launch line
 * |[
 * gst-launch-1.0 audiotestsrc ! "audio/x-raw,rate=48000" ! vorbisenc ! rtpvorbispay config-interval=1 ! rtpstreampay ! tcpserversink port=5678
 * 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
 * ]|
 *
 */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include "gstrtpstreamdepay.h"

GST_DEBUG_CATEGORY (gst_rtp_stream_depay_debug);
#define GST_CAT_DEFAULT gst_rtp_stream_depay_debug

static GstStaticPadTemplate src_template =
    GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("application/x-rtp; application/x-rtcp;"
        "application/x-srtp; application/x-srtcp")
    );

static GstStaticPadTemplate sink_template =
    GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("application/x-rtp-stream; application/x-rtcp-stream;"
        "application/x-srtp-stream; application/x-srtcp-stream")
    );

#define parent_class gst_rtp_stream_depay_parent_class
G_DEFINE_TYPE (GstRtpStreamDepay, gst_rtp_stream_depay, GST_TYPE_BASE_PARSE);

static gboolean gst_rtp_stream_depay_set_sink_caps (GstBaseParse * parse,
    GstCaps * caps);
static GstCaps *gst_rtp_stream_depay_get_sink_caps (GstBaseParse * parse,
    GstCaps * filter);
static GstFlowReturn gst_rtp_stream_depay_handle_frame (GstBaseParse * parse,
    GstBaseParseFrame * frame, gint * skipsize);

static gboolean gst_rtp_stream_depay_sink_activate (GstPad * pad,
    GstObject * parent);

static void
gst_rtp_stream_depay_class_init (GstRtpStreamDepayClass * klass)
{
  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
  GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);

  GST_DEBUG_CATEGORY_INIT (gst_rtp_stream_depay_debug, "rtpstreamdepay", 0,
      "RTP stream depayloader");

  gst_element_class_add_static_pad_template (gstelement_class, &src_template);
  gst_element_class_add_static_pad_template (gstelement_class, &sink_template);

  gst_element_class_set_static_metadata (gstelement_class,
      "RTP Stream Depayloading", "Codec/Depayloader/Network",
      "Depayloads RTP/RTCP packets for streaming protocols according to RFC4571",
      "Sebastian Dröge <sebastian@centricular.com>");

  parse_class->set_sink_caps =
      GST_DEBUG_FUNCPTR (gst_rtp_stream_depay_set_sink_caps);
  parse_class->get_sink_caps =
      GST_DEBUG_FUNCPTR (gst_rtp_stream_depay_get_sink_caps);
  parse_class->handle_frame =
      GST_DEBUG_FUNCPTR (gst_rtp_stream_depay_handle_frame);
}

static void
gst_rtp_stream_depay_init (GstRtpStreamDepay * self)
{
  gst_base_parse_set_min_frame_size (GST_BASE_PARSE (self), 2);

  /* Force activation in push mode. We need to get a caps event from upstream
   * to know the full RTP caps. */
  gst_pad_set_activate_function (GST_BASE_PARSE_SINK_PAD (self),
      gst_rtp_stream_depay_sink_activate);
}

static gboolean
gst_rtp_stream_depay_set_sink_caps (GstBaseParse * parse, GstCaps * caps)
{
  GstCaps *othercaps;
  GstStructure *structure;
  gboolean ret;

  othercaps = gst_caps_copy (caps);
  structure = gst_caps_get_structure (othercaps, 0);

  if (gst_structure_has_name (structure, "application/x-rtp-stream"))
    gst_structure_set_name (structure, "application/x-rtp");
  else if (gst_structure_has_name (structure, "application/x-rtcp-stream"))
    gst_structure_set_name (structure, "application/x-rtcp");
  else if (gst_structure_has_name (structure, "application/x-srtp-stream"))
    gst_structure_set_name (structure, "application/x-srtp");
  else
    gst_structure_set_name (structure, "application/x-srtcp");

  ret = gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), othercaps);
  gst_caps_unref (othercaps);

  return ret;
}

static GstCaps *
gst_rtp_stream_depay_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
{
  GstCaps *peerfilter = NULL, *peercaps, *templ;
  GstCaps *res;
  GstStructure *structure;
  guint i, n;

  if (filter) {
    peerfilter = gst_caps_copy (filter);
    n = gst_caps_get_size (peerfilter);
    for (i = 0; i < n; i++) {
      structure = gst_caps_get_structure (peerfilter, i);

      if (gst_structure_has_name (structure, "application/x-rtp-stream"))
        gst_structure_set_name (structure, "application/x-rtp");
      else if (gst_structure_has_name (structure, "application/x-rtcp-stream"))
        gst_structure_set_name (structure, "application/x-rtcp");
      else if (gst_structure_has_name (structure, "application/x-srtp-stream"))
        gst_structure_set_name (structure, "application/x-srtp");
      else
        gst_structure_set_name (structure, "application/x-srtcp");
    }
  }

  templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
  peercaps =
      gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), peerfilter);

  if (peercaps) {
    /* Rename structure names */
    peercaps = gst_caps_make_writable (peercaps);
    n = gst_caps_get_size (peercaps);
    for (i = 0; i < n; i++) {
      structure = gst_caps_get_structure (peercaps, i);

      if (gst_structure_has_name (structure, "application/x-rtp"))
        gst_structure_set_name (structure, "application/x-rtp-stream");
      else if (gst_structure_has_name (structure, "application/x-rtcp"))
        gst_structure_set_name (structure, "application/x-rtcp-stream");
      else if (gst_structure_has_name (structure, "application/x-srtp"))
        gst_structure_set_name (structure, "application/x-srtp-stream");
      else
        gst_structure_set_name (structure, "application/x-srtcp-stream");
    }

    res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
    gst_caps_unref (peercaps);
  } else {
    res = templ;
  }

  if (filter) {
    GstCaps *intersection;

    intersection =
        gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
    gst_caps_unref (res);
    res = intersection;

    gst_caps_unref (peerfilter);
  }

  return res;
}

static GstFlowReturn
gst_rtp_stream_depay_handle_frame (GstBaseParse * parse,
    GstBaseParseFrame * frame, gint * skipsize)
{
  gsize buf_size;
  guint16 size;

  if (gst_buffer_extract (frame->buffer, 0, &size, 2) != 2)
    return GST_FLOW_ERROR;

  size = GUINT16_FROM_BE (size);
  buf_size = gst_buffer_get_size (frame->buffer);

  /* Need more data */
  if (size + 2 > buf_size)
    return GST_FLOW_OK;

  frame->out_buffer =
      gst_buffer_copy_region (frame->buffer, GST_BUFFER_COPY_ALL, 2, size);

  return gst_base_parse_finish_frame (parse, frame, size + 2);
}

static gboolean
gst_rtp_stream_depay_sink_activate (GstPad * pad, GstObject * parent)
{
  return gst_pad_activate_mode (pad, GST_PAD_MODE_PUSH, TRUE);
}

gboolean
gst_rtp_stream_depay_plugin_init (GstPlugin * plugin)
{
  return gst_element_register (plugin, "rtpstreamdepay",
      GST_RANK_NONE, GST_TYPE_RTP_STREAM_DEPAY);
}