1 /* GStreamer
2 * Copyright (C) 2009 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 /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
21 * with newer GLib versions (>= 2.31.0) */
22 #define GLIB_DISABLE_DEPRECATION_WARNINGS
23
24 #include <string.h>
25 #include <math.h>
26
27 #include <gst/gst.h>
28
29 /*
30 * A simple RTP server
31 * sends the output of alsasrc as alaw encoded RTP on port 5002, RTCP is sent on
32 * port 5003. The destination is 127.0.0.1.
33 * the receiver RTCP reports are received on port 5007
34 *
35 * .-------. .-------. .-------. .----------. .-------.
36 * |alsasrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP
37 * | src->sink src->sink src->send_rtp send_rtp->sink | port=5002
38 * '-------' '-------' '-------' | | '-------'
39 * | |
40 * | | .-------.
41 * | | |udpsink| RTCP
42 * | send_rtcp->sink | port=5003
43 * .-------. | | '-------' sync=false
44 * RTCP |udpsrc | | | async=false
45 * port=5007 | src->recv_rtcp |
46 * '-------' '----------'
47 */
48
49 /* change this to send the RTP data and RTCP to another host */
50 #define DEST_HOST "127.0.0.1"
51
52 /* #define AUDIO_SRC "alsasrc" */
53 #define AUDIO_SRC "audiotestsrc"
54
55 /* the encoder and payloader elements */
56 #define AUDIO_ENC "alawenc"
57 #define AUDIO_PAY "rtppcmapay"
58
59 /* print the stats of a source */
60 static void
print_source_stats(GObject * source)61 print_source_stats (GObject * source)
62 {
63 GstStructure *stats;
64 gchar *str;
65
66 /* get the source stats */
67 g_object_get (source, "stats", &stats, NULL);
68
69 /* simply dump the stats structure */
70 str = gst_structure_to_string (stats);
71 g_print ("source stats: %s\n", str);
72
73 gst_structure_free (stats);
74 g_free (str);
75 }
76
77 /* this function is called every second and dumps the RTP manager stats */
78 static gboolean
print_stats(GstElement * rtpbin)79 print_stats (GstElement * rtpbin)
80 {
81 GObject *session;
82 GValueArray *arr;
83 GValue *val;
84 guint i;
85
86 g_print ("***********************************\n");
87
88 /* get session 0 */
89 g_signal_emit_by_name (rtpbin, "get-internal-session", 0, &session);
90
91 /* print all the sources in the session, this includes the internal source */
92 g_object_get (session, "sources", &arr, NULL);
93
94 for (i = 0; i < arr->n_values; i++) {
95 GObject *source;
96
97 val = g_value_array_get_nth (arr, i);
98 source = g_value_get_object (val);
99
100 print_source_stats (source);
101 }
102 g_value_array_free (arr);
103
104 g_object_unref (session);
105
106 return TRUE;
107 }
108
109 /* build a pipeline equivalent to:
110 *
111 * gst-launch-1.0 -v rtpbin name=rtpbin \
112 * $AUDIO_SRC ! audioconvert ! audioresample ! $AUDIO_ENC ! $AUDIO_PAY ! rtpbin.send_rtp_sink_0 \
113 * rtpbin.send_rtp_src_0 ! udpsink port=5002 host=$DEST \
114 * rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=$DEST sync=false async=false \
115 * udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0
116 */
117 int
main(int argc,char * argv[])118 main (int argc, char *argv[])
119 {
120 GstElement *audiosrc, *audioconv, *audiores, *audioenc, *audiopay;
121 GstElement *rtpbin, *rtpsink, *rtcpsink, *rtcpsrc;
122 GstElement *pipeline;
123 GMainLoop *loop;
124 GstPad *srcpad, *sinkpad;
125
126 /* always init first */
127 gst_init (&argc, &argv);
128
129 /* the pipeline to hold everything */
130 pipeline = gst_pipeline_new (NULL);
131 g_assert (pipeline);
132
133 /* the audio capture and format conversion */
134 audiosrc = gst_element_factory_make (AUDIO_SRC, "audiosrc");
135 g_assert (audiosrc);
136 audioconv = gst_element_factory_make ("audioconvert", "audioconv");
137 g_assert (audioconv);
138 audiores = gst_element_factory_make ("audioresample", "audiores");
139 g_assert (audiores);
140 /* the encoding and payloading */
141 audioenc = gst_element_factory_make (AUDIO_ENC, "audioenc");
142 g_assert (audioenc);
143 audiopay = gst_element_factory_make (AUDIO_PAY, "audiopay");
144 g_assert (audiopay);
145
146 /* add capture and payloading to the pipeline and link */
147 gst_bin_add_many (GST_BIN (pipeline), audiosrc, audioconv, audiores,
148 audioenc, audiopay, NULL);
149
150 if (!gst_element_link_many (audiosrc, audioconv, audiores, audioenc,
151 audiopay, NULL)) {
152 g_error ("Failed to link audiosrc, audioconv, audioresample, "
153 "audio encoder and audio payloader");
154 }
155
156 /* the rtpbin element */
157 rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
158 g_assert (rtpbin);
159
160 gst_bin_add (GST_BIN (pipeline), rtpbin);
161
162 /* the udp sinks and source we will use for RTP and RTCP */
163 rtpsink = gst_element_factory_make ("udpsink", "rtpsink");
164 g_assert (rtpsink);
165 g_object_set (rtpsink, "port", 5002, "host", DEST_HOST, NULL);
166
167 rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink");
168 g_assert (rtcpsink);
169 g_object_set (rtcpsink, "port", 5003, "host", DEST_HOST, NULL);
170 /* no need for synchronisation or preroll on the RTCP sink */
171 g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL);
172
173 rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc");
174 g_assert (rtcpsrc);
175 g_object_set (rtcpsrc, "port", 5007, NULL);
176
177 gst_bin_add_many (GST_BIN (pipeline), rtpsink, rtcpsink, rtcpsrc, NULL);
178
179 /* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */
180 sinkpad = gst_element_request_pad_simple (rtpbin, "send_rtp_sink_0");
181 srcpad = gst_element_get_static_pad (audiopay, "src");
182 if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
183 g_error ("Failed to link audio payloader to rtpbin");
184 gst_object_unref (srcpad);
185
186 /* get the RTP srcpad that was created when we requested the sinkpad above and
187 * link it to the rtpsink sinkpad*/
188 srcpad = gst_element_get_static_pad (rtpbin, "send_rtp_src_0");
189 sinkpad = gst_element_get_static_pad (rtpsink, "sink");
190 if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
191 g_error ("Failed to link rtpbin to rtpsink");
192 gst_object_unref (srcpad);
193 gst_object_unref (sinkpad);
194
195 /* get an RTCP srcpad for sending RTCP to the receiver */
196 srcpad = gst_element_request_pad_simple (rtpbin, "send_rtcp_src_0");
197 sinkpad = gst_element_get_static_pad (rtcpsink, "sink");
198 if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
199 g_error ("Failed to link rtpbin to rtcpsink");
200 gst_object_unref (sinkpad);
201
202 /* we also want to receive RTCP, request an RTCP sinkpad for session 0 and
203 * link it to the srcpad of the udpsrc for RTCP */
204 srcpad = gst_element_get_static_pad (rtcpsrc, "src");
205 sinkpad = gst_element_request_pad_simple (rtpbin, "recv_rtcp_sink_0");
206 if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
207 g_error ("Failed to link rtcpsrc to rtpbin");
208 gst_object_unref (srcpad);
209
210 /* set the pipeline to playing */
211 g_print ("starting sender pipeline\n");
212 gst_element_set_state (pipeline, GST_STATE_PLAYING);
213
214 /* print stats every second */
215 g_timeout_add_seconds (1, (GSourceFunc) print_stats, rtpbin);
216
217 /* we need to run a GLib main loop to get the messages */
218 loop = g_main_loop_new (NULL, FALSE);
219 g_main_loop_run (loop);
220
221 g_print ("stopping sender pipeline\n");
222 gst_element_set_state (pipeline, GST_STATE_NULL);
223
224 return 0;
225 }
226