gstreamer/tests/examples/webrtc/webrtcrenego.c

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#include <gst/gst.h>
#include <gst/sdp/sdp.h>
#include <gst/webrtc/webrtc.h>
#include <string.h>
static GMainLoop *loop;
static GstElement *pipe1, *webrtc1, *webrtc2, *extra_src;
static GstBus *bus1;
#define SEND_SRC(pattern) "videotestsrc is-live=true pattern=" pattern " ! timeoverlay ! queue ! vp8enc ! rtpvp8pay ! queue ! " \
"capsfilter caps=application/x-rtp,media=video,payload=96,encoding-name=VP8"
static void
_element_message (GstElement * parent, GstMessage * msg)
{
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:{
GstElement *receive, *webrtc;
GstPad *pad, *peer;
g_print ("Got element EOS message from %s parent %s\n",
GST_OBJECT_NAME (msg->src), GST_OBJECT_NAME (parent));
receive = GST_ELEMENT (msg->src);
pad = gst_element_get_static_pad (receive, "sink");
peer = gst_pad_get_peer (pad);
webrtc = GST_ELEMENT (gst_pad_get_parent (peer));
gst_bin_remove (GST_BIN (pipe1), receive);
gst_pad_unlink (peer, pad);
gst_element_release_request_pad (webrtc, peer);
gst_object_unref (pad);
gst_object_unref (peer);
gst_element_set_state (receive, GST_STATE_NULL);
break;
}
default:
break;
}
}
static gboolean
_bus_watch (GstBus * bus, GstMessage * msg, GstElement * pipe)
{
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_STATE_CHANGED:
if (GST_ELEMENT (msg->src) == pipe) {
GstState old, new, pending;
gst_message_parse_state_changed (msg, &old, &new, &pending);
{
gchar *dump_name = g_strconcat ("state_changed-",
gst_element_state_get_name (old), "_",
gst_element_state_get_name (new), NULL);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (msg->src),
GST_DEBUG_GRAPH_SHOW_ALL, dump_name);
g_free (dump_name);
}
}
break;
case GST_MESSAGE_ERROR:{
GError *err = NULL;
gchar *dbg_info = NULL;
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe),
GST_DEBUG_GRAPH_SHOW_ALL, "error");
gst_message_parse_error (msg, &err, &dbg_info);
g_printerr ("ERROR from element %s: %s\n",
GST_OBJECT_NAME (msg->src), err->message);
g_printerr ("Debugging info: %s\n", (dbg_info) ? dbg_info : "none");
g_error_free (err);
g_free (dbg_info);
g_main_loop_quit (loop);
break;
}
case GST_MESSAGE_EOS:{
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe),
GST_DEBUG_GRAPH_SHOW_ALL, "eos");
g_print ("EOS received\n");
g_main_loop_quit (loop);
break;
}
case GST_MESSAGE_ELEMENT:{
const GstStructure *s = gst_message_get_structure (msg);
if (g_strcmp0 (gst_structure_get_name (s), "GstBinForwarded") == 0) {
GstMessage *sub_msg;
gst_structure_get (s, "message", GST_TYPE_MESSAGE, &sub_msg, NULL);
_element_message (GST_ELEMENT (msg->src), sub_msg);
gst_message_unref (sub_msg);
}
break;
}
default:
break;
}
return TRUE;
}
static void
_webrtc_pad_added (GstElement * webrtc, GstPad * new_pad, GstElement * pipe)
{
GstElement *out;
GstPad *sink;
if (GST_PAD_DIRECTION (new_pad) != GST_PAD_SRC)
return;
out = gst_parse_bin_from_description ("queue ! rtpvp8depay ! vp8dec ! "
"videoconvert ! queue ! xvimagesink", TRUE, NULL);
gst_bin_add (GST_BIN (pipe), out);
gst_element_sync_state_with_parent (out);
sink = out->sinkpads->data;
gst_pad_link (new_pad, sink);
}
static void
_on_answer_received (GstPromise * promise, gpointer user_data)
{
GstWebRTCSessionDescription *answer = NULL;
const GstStructure *reply;
gchar *desc;
g_assert (gst_promise_wait (promise) == GST_PROMISE_RESULT_REPLIED);
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "answer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL);
gst_promise_unref (promise);
desc = gst_sdp_message_as_text (answer->sdp);
g_print ("Created answer:\n%s\n", desc);
g_free (desc);
/* this is one way to tell webrtcbin that we don't want to be notified when
* this task is complete: set a NULL promise */
g_signal_emit_by_name (webrtc1, "set-remote-description", answer, NULL);
/* this is another way to tell webrtcbin that we don't want to be notified
* when this task is complete: interrupt the promise */
promise = gst_promise_new ();
g_signal_emit_by_name (webrtc2, "set-local-description", answer, promise);
gst_promise_interrupt (promise);
gst_promise_unref (promise);
gst_webrtc_session_description_free (answer);
}
static void
_on_offer_received (GstPromise * promise, gpointer user_data)
{
GstWebRTCSessionDescription *offer = NULL;
const GstStructure *reply;
gchar *desc;
g_assert (gst_promise_wait (promise) == GST_PROMISE_RESULT_REPLIED);
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "offer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
gst_promise_unref (promise);
desc = gst_sdp_message_as_text (offer->sdp);
g_print ("Created offer:\n%s\n", desc);
g_free (desc);
g_signal_emit_by_name (webrtc1, "set-local-description", offer, NULL);
g_signal_emit_by_name (webrtc2, "set-remote-description", offer, NULL);
promise = gst_promise_new_with_change_func (_on_answer_received, user_data,
NULL);
g_signal_emit_by_name (webrtc2, "create-answer", NULL, promise);
gst_webrtc_session_description_free (offer);
}
static void
_on_negotiation_needed (GstElement * element, gpointer user_data)
{
GstPromise *promise;
promise = gst_promise_new_with_change_func (_on_offer_received, user_data,
NULL);
g_signal_emit_by_name (webrtc1, "create-offer", NULL, promise);
}
static void
_on_ice_candidate (GstElement * webrtc, guint mlineindex, gchar * candidate,
GstElement * other)
{
g_signal_emit_by_name (other, "add-ice-candidate", mlineindex, candidate);
}
static gboolean
stream_change (gpointer data)
{
if (!extra_src) {
g_print ("Adding extra stream\n");
extra_src =
gst_parse_bin_from_description (SEND_SRC ("circular"), TRUE, NULL);
gst_element_set_locked_state (extra_src, TRUE);
gst_bin_add (GST_BIN (pipe1), extra_src);
gst_element_link (extra_src, webrtc1);
gst_element_set_locked_state (extra_src, FALSE);
gst_element_sync_state_with_parent (extra_src);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe1),
GST_DEBUG_GRAPH_SHOW_ALL, "add");
} else {
GstPad *pad, *peer;
GstWebRTCRTPTransceiver *transceiver;
g_print ("Removing extra stream\n");
pad = gst_element_get_static_pad (extra_src, "src");
peer = gst_pad_get_peer (pad);
gst_element_send_event (extra_src, gst_event_new_eos ());
g_object_get (peer, "transceiver", &transceiver, NULL);
g_object_set (transceiver, "direction",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE, NULL);
gst_element_set_locked_state (extra_src, TRUE);
gst_element_set_state (extra_src, GST_STATE_NULL);
gst_pad_unlink (pad, peer);
gst_element_release_request_pad (webrtc1, peer);
gst_object_unref (peer);
gst_object_unref (pad);
gst_bin_remove (GST_BIN (pipe1), extra_src);
extra_src = NULL;
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipe1),
GST_DEBUG_GRAPH_SHOW_ALL, "remove");
}
return G_SOURCE_CONTINUE;
}
int
main (int argc, char *argv[])
{
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
pipe1 = gst_parse_launch (SEND_SRC ("smpte")
" ! webrtcbin name=smpte bundle-policy=max-bundle " SEND_SRC ("ball")
" ! webrtcbin name=ball bundle-policy=max-bundle", NULL);
g_object_set (pipe1, "message-forward", TRUE, NULL);
bus1 = gst_pipeline_get_bus (GST_PIPELINE (pipe1));
gst_bus_add_watch (bus1, (GstBusFunc) _bus_watch, pipe1);
webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "smpte");
g_signal_connect (webrtc1, "on-negotiation-needed",
G_CALLBACK (_on_negotiation_needed), NULL);
g_signal_connect (webrtc1, "pad-added", G_CALLBACK (_webrtc_pad_added),
pipe1);
webrtc2 = gst_bin_get_by_name (GST_BIN (pipe1), "ball");
g_signal_connect (webrtc2, "pad-added", G_CALLBACK (_webrtc_pad_added),
pipe1);
g_signal_connect (webrtc1, "on-ice-candidate",
G_CALLBACK (_on_ice_candidate), webrtc2);
g_signal_connect (webrtc2, "on-ice-candidate",
G_CALLBACK (_on_ice_candidate), webrtc1);
g_print ("Starting pipeline\n");
gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_PLAYING);
g_timeout_add_seconds (5, stream_change, NULL);
g_main_loop_run (loop);
gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_NULL);
g_print ("Pipeline stopped\n");
gst_object_unref (webrtc1);
gst_object_unref (webrtc2);
gst_bus_remove_watch (bus1);
gst_object_unref (bus1);
gst_object_unref (pipe1);
gst_deinit ();
return 0;
}