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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;
static GstBus *bus1;
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;
}
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 ("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);
g_signal_emit_by_name (webrtc1, "set-remote-description", answer, NULL);
g_signal_emit_by_name (webrtc2, "set-local-description", answer, NULL);
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 void
_on_new_transceiver (GstElement * webrtc, GstWebRTCRTPTransceiver * trans)
{
/* If we expected more than one transceiver, we would take a look at
* trans->mline, and compare it with webrtcbin's local description */
g_object_set (trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED, NULL);
}
static void
add_fec_to_offer (GstElement * webrtc)
{
GstWebRTCRTPTransceiver *trans;
GArray *transceivers;
/* A transceiver has already been created when a sink pad was
* requested on the sending webrtcbin */
g_signal_emit_by_name (webrtc, "get-transceivers", &transceivers);
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);
g_object_set (trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED,
"fec-percentage", 100, NULL);
g_array_unref (transceivers);
}
int
main (int argc, char *argv[])
{
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
pipe1 =
gst_parse_launch
("videotestsrc pattern=ball ! video/x-raw ! queue ! vp8enc ! rtpvp8pay ! queue ! "
"application/x-rtp,media=video,payload=96,encoding-name=VP8 ! "
"webrtcbin name=send webrtcbin name=recv", 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), "send");
g_signal_connect (webrtc1, "on-negotiation-needed",
G_CALLBACK (_on_negotiation_needed), NULL);
add_fec_to_offer (webrtc1);
webrtc2 = gst_bin_get_by_name (GST_BIN (pipe1), "recv");
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_signal_connect (webrtc2, "on-new-transceiver",
G_CALLBACK (_on_new_transceiver), NULL);
g_print ("Starting pipeline\n");
gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_PLAYING);
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;
}
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