gstreamer/src/webrtc/simple/webrtcbidirectional.c
#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);
/* 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, NULL);
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);
}
int main(int argc, char *argv[])
{
gst_init(&argc, &argv);
loop = g_main_loop_new(NULL, FALSE);
pipe1 =
gst_parse_launch(
"videotestsrc pattern=white ! timeoverlay valignment=3 halignment=4 time-mode=2 xpos=0 ypos=0 color=4278190080 font-desc=\"Sans 48\" draw-shadow=false draw-outline=true outline-color=4278190080 ! queue ! vp8enc ! rtpvp8pay ! queue ! "
"application/x-rtp,media=video,payload=96,encoding-name=VP8 ! "
"webrtcbin name=smpte videotestsrc pattern=ball ! queue ! vp8enc ! rtpvp8pay ! queue ! "
"application/x-rtp,media=video,payload=96,encoding-name=VP8 ! webrtcbin name=ball",
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_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;
}