2 #include <gst/sdp/sdp.h>
3 #include <gst/webrtc/webrtc.h>
7 static GMainLoop *loop;
8 static GstElement *pipe1, *webrtc1, *webrtc2;
12 _bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe)
14 switch (GST_MESSAGE_TYPE (msg)) {
15 case GST_MESSAGE_STATE_CHANGED:
16 if (GST_ELEMENT (msg->src) == pipe) {
17 GstState old, new, pending;
19 gst_message_parse_state_changed (msg, &old, &new, &pending);
22 gchar *dump_name = g_strconcat ("state_changed-",
23 gst_element_state_get_name (old), "_",
24 gst_element_state_get_name (new), NULL);
25 GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (msg->src),
26 GST_DEBUG_GRAPH_SHOW_ALL, dump_name);
31 case GST_MESSAGE_ERROR:{
33 gchar *dbg_info = NULL;
35 GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe),
36 GST_DEBUG_GRAPH_SHOW_ALL, "error");
38 gst_message_parse_error (msg, &err, &dbg_info);
39 g_printerr ("ERROR from element %s: %s\n",
40 GST_OBJECT_NAME (msg->src), err->message);
41 g_printerr ("Debugging info: %s\n", (dbg_info) ? dbg_info : "none");
44 g_main_loop_quit (loop);
47 case GST_MESSAGE_EOS:{
48 GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe),
49 GST_DEBUG_GRAPH_SHOW_ALL, "eos");
50 g_print ("EOS received\n");
51 g_main_loop_quit (loop);
62 _webrtc_pad_added (GstElement * webrtc, GstPad * new_pad, GstElement * pipe)
64 GstElement *out = NULL;
68 const gchar *encoding_name;
70 if (GST_PAD_DIRECTION (new_pad) != GST_PAD_SRC)
73 caps = gst_pad_get_current_caps (new_pad);
75 caps = gst_pad_query_caps (new_pad, NULL);
76 GST_ERROR_OBJECT (new_pad, "caps %" GST_PTR_FORMAT, caps);
77 g_assert (gst_caps_is_fixed (caps));
78 s = gst_caps_get_structure (caps, 0);
79 encoding_name = gst_structure_get_string (s, "encoding-name");
80 if (g_strcmp0 (encoding_name, "VP8") == 0) {
81 out = gst_parse_bin_from_description ("rtpvp8depay ! vp8dec ! "
82 "videoconvert ! queue ! xvimagesink sync=false", TRUE, NULL);
83 } else if (g_strcmp0 (encoding_name, "OPUS") == 0) {
84 out = gst_parse_bin_from_description ("rtpopusdepay ! opusdec ! "
85 "audioconvert ! audioresample ! audiorate ! queue ! autoaudiosink",
88 g_critical ("Unknown encoding name %s", encoding_name);
89 g_assert_not_reached ();
91 gst_bin_add (GST_BIN (pipe), out);
92 gst_element_sync_state_with_parent (out);
93 sink = out->sinkpads->data;
95 gst_pad_link (new_pad, sink);
97 gst_caps_unref (caps);
101 _on_answer_received (GstPromise * promise, gpointer user_data)
103 GstWebRTCSessionDescription *answer = NULL;
104 const GstStructure *reply;
107 g_assert (gst_promise_wait (promise) == GST_PROMISE_RESULT_REPLIED);
108 reply = gst_promise_get_reply (promise);
109 gst_structure_get (reply, "answer",
110 GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL);
111 gst_promise_unref (promise);
112 desc = gst_sdp_message_as_text (answer->sdp);
113 g_print ("Created answer:\n%s\n", desc);
116 g_signal_emit_by_name (webrtc1, "set-remote-description", answer, NULL);
117 g_signal_emit_by_name (webrtc2, "set-local-description", answer, NULL);
119 gst_webrtc_session_description_free (answer);
123 _on_offer_received (GstPromise * promise, gpointer user_data)
125 GstWebRTCSessionDescription *offer = NULL;
126 const GstStructure *reply;
129 g_assert (gst_promise_wait (promise) == GST_PROMISE_RESULT_REPLIED);
130 reply = gst_promise_get_reply (promise);
131 gst_structure_get (reply, "offer",
132 GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
133 gst_promise_unref (promise);
134 desc = gst_sdp_message_as_text (offer->sdp);
135 g_print ("Created offer:\n%s\n", desc);
138 g_signal_emit_by_name (webrtc1, "set-local-description", offer, NULL);
139 g_signal_emit_by_name (webrtc2, "set-remote-description", offer, NULL);
141 promise = gst_promise_new_with_change_func (_on_answer_received, user_data,
143 g_signal_emit_by_name (webrtc2, "create-answer", NULL, promise);
145 gst_webrtc_session_description_free (offer);
149 _on_negotiation_needed (GstElement * element, gpointer user_data)
153 promise = gst_promise_new_with_change_func (_on_offer_received, user_data,
155 g_signal_emit_by_name (webrtc1, "create-offer", NULL, promise);
159 _on_ice_candidate (GstElement * webrtc, guint mlineindex, gchar * candidate,
162 g_signal_emit_by_name (other, "add-ice-candidate", mlineindex, candidate);
166 main (int argc, char *argv[])
168 gst_init (&argc, &argv);
170 loop = g_main_loop_new (NULL, FALSE);
172 gst_parse_launch ("webrtcbin name=smpte webrtcbin name=ball "
173 "videotestsrc pattern=smpte ! queue ! vp8enc ! rtpvp8pay ! queue ! "
174 "application/x-rtp,media=video,payload=96,encoding-name=VP8 ! smpte.sink_0 "
175 "audiotestsrc ! opusenc ! rtpopuspay ! queue ! "
176 "application/x-rtp,media=audio,payload=97,encoding-name=OPUS ! smpte.sink_1 "
177 "videotestsrc pattern=ball ! queue ! vp8enc ! rtpvp8pay ! queue ! "
178 "application/x-rtp,media=video,payload=96,encoding-name=VP8 ! ball.sink_1 "
179 "audiotestsrc wave=saw ! opusenc ! rtpopuspay ! queue ! "
180 "application/x-rtp,media=audio,payload=97,encoding-name=OPUS ! ball.sink_0 ",
182 bus1 = gst_pipeline_get_bus (GST_PIPELINE (pipe1));
183 gst_bus_add_watch (bus1, (GstBusFunc) _bus_watch, pipe1);
185 webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "smpte");
186 g_signal_connect (webrtc1, "on-negotiation-needed",
187 G_CALLBACK (_on_negotiation_needed), NULL);
188 g_signal_connect (webrtc1, "pad-added", G_CALLBACK (_webrtc_pad_added),
190 webrtc2 = gst_bin_get_by_name (GST_BIN (pipe1), "ball");
191 g_signal_connect (webrtc2, "pad-added", G_CALLBACK (_webrtc_pad_added),
193 g_signal_connect (webrtc1, "on-ice-candidate",
194 G_CALLBACK (_on_ice_candidate), webrtc2);
195 g_signal_connect (webrtc2, "on-ice-candidate",
196 G_CALLBACK (_on_ice_candidate), webrtc1);
198 g_print ("Starting pipeline\n");
199 gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_PLAYING);
201 g_main_loop_run (loop);
203 gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_NULL);
204 g_print ("Pipeline stopped\n");
206 gst_object_unref (webrtc1);
207 gst_object_unref (webrtc2);
208 gst_bus_remove_watch (bus1);
209 gst_object_unref (bus1);
210 gst_object_unref (pipe1);