gstreamer/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
Nirbheek Chauhan 033a71e405 webrtc examples: Use webrtc.gstreamer.net
Actually just a CNAME to webrtc.nirbheek.in for now, but it allows
replacement / hosting without my involvement, so reduces the bus
factor.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3802>
2023-02-04 13:37:02 +00:00

1035 lines
30 KiB
C

/*
* Demo gstreamer app for negotiating and streaming a sendrecv audio-only webrtc
* stream to all the peers in a multiparty room.
*
* gcc mp-webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o mp-webrtc-sendrecv
*
* Author: Nirbheek Chauhan <nirbheek@centricular.com>
*/
#include <gst/gst.h>
#include <gst/sdp/sdp.h>
#define GST_USE_UNSTABLE_API
#include <gst/webrtc/webrtc.h>
/* For signalling */
#include <libsoup/soup.h>
#include <json-glib/json-glib.h>
#include <string.h>
enum AppState
{
APP_STATE_UNKNOWN = 0,
APP_STATE_ERROR = 1, /* generic error */
SERVER_CONNECTING = 1000,
SERVER_CONNECTION_ERROR,
SERVER_CONNECTED, /* Ready to register */
SERVER_REGISTERING = 2000,
SERVER_REGISTRATION_ERROR,
SERVER_REGISTERED, /* Ready to call a peer */
SERVER_CLOSED, /* server connection closed by us or the server */
ROOM_JOINING = 3000,
ROOM_JOIN_ERROR,
ROOM_JOINED,
ROOM_CALL_NEGOTIATING = 4000, /* negotiating with some or all peers */
ROOM_CALL_OFFERING, /* when we're the one sending the offer */
ROOM_CALL_ANSWERING, /* when we're the one answering an offer */
ROOM_CALL_STARTED, /* in a call with some or all peers */
ROOM_CALL_STOPPING,
ROOM_CALL_STOPPED,
ROOM_CALL_ERROR,
};
static GMainLoop *loop;
static GstElement *pipeline;
static GList *peers;
static SoupWebsocketConnection *ws_conn = NULL;
static enum AppState app_state = 0;
static const gchar *default_server_url = "wss://webrtc.gstreamer.net:8443";
static gchar *server_url = NULL;
static gchar *local_id = NULL;
static gchar *room_id = NULL;
static gboolean strict_ssl = TRUE;
static GOptionEntry entries[] = {
{"name", 0, 0, G_OPTION_ARG_STRING, &local_id,
"Name we will send to the server", "ID"},
{"room-id", 0, 0, G_OPTION_ARG_STRING, &room_id,
"Room name to join or create", "ID"},
{"server", 0, 0, G_OPTION_ARG_STRING, &server_url,
"Signalling server to connect to", "URL"},
{NULL}
};
static gint
compare_str_glist (gconstpointer a, gconstpointer b)
{
return g_strcmp0 (a, b);
}
static const gchar *
find_peer_from_list (const gchar * peer_id)
{
return (g_list_find_custom (peers, peer_id, compare_str_glist))->data;
}
static gboolean
cleanup_and_quit_loop (const gchar * msg, enum AppState state)
{
if (msg)
gst_printerr ("%s\n", msg);
if (state > 0)
app_state = state;
if (ws_conn) {
if (soup_websocket_connection_get_state (ws_conn) ==
SOUP_WEBSOCKET_STATE_OPEN)
/* This will call us again */
soup_websocket_connection_close (ws_conn, 1000, "");
else
g_object_unref (ws_conn);
}
if (loop) {
g_main_loop_quit (loop);
loop = NULL;
}
/* To allow usage as a GSourceFunc */
return G_SOURCE_REMOVE;
}
static gchar *
get_string_from_json_object (JsonObject * object)
{
JsonNode *root;
JsonGenerator *generator;
gchar *text;
/* Make it the root node */
root = json_node_init_object (json_node_alloc (), object);
generator = json_generator_new ();
json_generator_set_root (generator, root);
text = json_generator_to_data (generator, NULL);
/* Release everything */
g_object_unref (generator);
json_node_free (root);
return text;
}
static gboolean
bus_watch_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
GstPipeline *pipeline = user_data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
{
GError *error = NULL;
gchar *debug = NULL;
gst_message_parse_error (message, &error, &debug);
g_error ("Error on bus: %s (debug: %s)", error->message, debug);
g_error_free (error);
g_free (debug);
break;
}
case GST_MESSAGE_WARNING:
{
GError *error = NULL;
gchar *debug = NULL;
gst_message_parse_warning (message, &error, &debug);
g_warning ("Warning on bus: %s (debug: %s)", error->message, debug);
g_error_free (error);
g_free (debug);
break;
}
case GST_MESSAGE_LATENCY:
gst_bin_recalculate_latency (GST_BIN (pipeline));
break;
default:
break;
}
return G_SOURCE_CONTINUE;
}
static void
handle_media_stream (GstPad * pad, GstElement * pipe, const char *convert_name,
const char *sink_name)
{
GstPad *qpad;
GstElement *q, *conv, *sink;
GstPadLinkReturn ret;
q = gst_element_factory_make ("queue", NULL);
g_assert_nonnull (q);
conv = gst_element_factory_make (convert_name, NULL);
g_assert_nonnull (conv);
sink = gst_element_factory_make (sink_name, NULL);
g_assert_nonnull (sink);
gst_bin_add_many (GST_BIN (pipe), q, conv, sink, NULL);
gst_element_sync_state_with_parent (q);
gst_element_sync_state_with_parent (conv);
gst_element_sync_state_with_parent (sink);
gst_element_link_many (q, conv, sink, NULL);
qpad = gst_element_get_static_pad (q, "sink");
ret = gst_pad_link (pad, qpad);
g_assert_cmpint (ret, ==, GST_PAD_LINK_OK);
}
static void
on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
GstElement * pipe)
{
GstCaps *caps;
const gchar *name;
if (!gst_pad_has_current_caps (pad)) {
gst_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
GST_PAD_NAME (pad));
return;
}
caps = gst_pad_get_current_caps (pad);
name = gst_structure_get_name (gst_caps_get_structure (caps, 0));
if (g_str_has_prefix (name, "video")) {
handle_media_stream (pad, pipe, "videoconvert", "autovideosink");
} else if (g_str_has_prefix (name, "audio")) {
handle_media_stream (pad, pipe, "audioconvert", "autoaudiosink");
} else {
gst_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad));
}
}
static void
on_incoming_stream (GstElement * webrtc, GstPad * pad, GstElement * pipe)
{
GstElement *decodebin;
GstPad *sinkpad;
if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC)
return;
decodebin = gst_element_factory_make ("decodebin", NULL);
g_signal_connect (decodebin, "pad-added",
G_CALLBACK (on_incoming_decodebin_stream), pipe);
gst_bin_add (GST_BIN (pipe), decodebin);
gst_element_sync_state_with_parent (decodebin);
sinkpad = gst_element_get_static_pad (decodebin, "sink");
gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
}
static void
send_room_peer_msg (const gchar * text, const gchar * peer_id)
{
gchar *msg;
msg = g_strdup_printf ("ROOM_PEER_MSG %s %s", peer_id, text);
soup_websocket_connection_send_text (ws_conn, msg);
g_free (msg);
}
static void
send_ice_candidate_message (GstElement * webrtc G_GNUC_UNUSED, guint mlineindex,
gchar * candidate, const gchar * peer_id)
{
gchar *text;
JsonObject *ice, *msg;
if (app_state < ROOM_CALL_OFFERING) {
cleanup_and_quit_loop ("Can't send ICE, not in call", APP_STATE_ERROR);
return;
}
ice = json_object_new ();
json_object_set_string_member (ice, "candidate", candidate);
json_object_set_int_member (ice, "sdpMLineIndex", mlineindex);
msg = json_object_new ();
json_object_set_object_member (msg, "ice", ice);
text = get_string_from_json_object (msg);
json_object_unref (msg);
send_room_peer_msg (text, peer_id);
g_free (text);
}
static void
send_room_peer_sdp (GstWebRTCSessionDescription * desc, const gchar * peer_id)
{
JsonObject *msg, *sdp;
gchar *text, *sdptype, *sdptext;
g_assert_cmpint (app_state, >=, ROOM_CALL_OFFERING);
if (desc->type == GST_WEBRTC_SDP_TYPE_OFFER)
sdptype = "offer";
else if (desc->type == GST_WEBRTC_SDP_TYPE_ANSWER)
sdptype = "answer";
else
g_assert_not_reached ();
text = gst_sdp_message_as_text (desc->sdp);
gst_print ("Sending sdp %s to %s:\n%s\n", sdptype, peer_id, text);
sdp = json_object_new ();
json_object_set_string_member (sdp, "type", sdptype);
json_object_set_string_member (sdp, "sdp", text);
g_free (text);
msg = json_object_new ();
json_object_set_object_member (msg, "sdp", sdp);
sdptext = get_string_from_json_object (msg);
json_object_unref (msg);
send_room_peer_msg (sdptext, peer_id);
g_free (sdptext);
}
/* Offer created by our pipeline, to be sent to the peer */
static void
on_offer_created (GstPromise * promise, const gchar * peer_id)
{
GstElement *webrtc;
GstWebRTCSessionDescription *offer;
const GstStructure *reply;
g_assert_cmpint (app_state, ==, ROOM_CALL_OFFERING);
g_assert_cmpint (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);
promise = gst_promise_new ();
webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id);
g_assert_nonnull (webrtc);
g_signal_emit_by_name (webrtc, "set-local-description", offer, promise);
gst_promise_interrupt (promise);
gst_promise_unref (promise);
/* Send offer to peer */
send_room_peer_sdp (offer, peer_id);
gst_webrtc_session_description_free (offer);
gst_object_unref (webrtc);
}
static void
on_negotiation_needed (GstElement * webrtc, const gchar * peer_id)
{
GstPromise *promise;
app_state = ROOM_CALL_OFFERING;
promise = gst_promise_new_with_change_func (
(GstPromiseChangeFunc) on_offer_created, (gpointer) peer_id, NULL);
g_signal_emit_by_name (webrtc, "create-offer", NULL, promise);
}
static void
remove_peer_from_pipeline (const gchar * peer_id)
{
gchar *qname;
GstPad *srcpad, *sinkpad;
GstElement *webrtc, *q, *tee;
webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id);
if (!webrtc)
return;
gst_bin_remove (GST_BIN (pipeline), webrtc);
gst_element_set_state (GST_ELEMENT (webrtc), GST_STATE_NULL);
gst_object_unref (webrtc);
qname = g_strdup_printf ("queue-%s", peer_id);
q = gst_bin_get_by_name (GST_BIN (pipeline), qname);
g_free (qname);
sinkpad = gst_element_get_static_pad (q, "sink");
g_assert_nonnull (sinkpad);
srcpad = gst_pad_get_peer (sinkpad);
g_assert_nonnull (srcpad);
gst_object_unref (sinkpad);
gst_bin_remove (GST_BIN (pipeline), q);
gst_element_set_state (GST_ELEMENT (q), GST_STATE_NULL);
gst_object_unref (q);
tee = gst_bin_get_by_name (GST_BIN (pipeline), "audiotee");
g_assert_nonnull (tee);
gst_element_release_request_pad (tee, srcpad);
gst_object_unref (srcpad);
gst_object_unref (tee);
}
static void
add_peer_to_pipeline (const gchar * peer_id, gboolean offer)
{
int ret;
gchar *tmp;
GstElement *tee, *webrtc, *q;
GstPad *srcpad, *sinkpad;
tmp = g_strdup_printf ("queue-%s", peer_id);
q = gst_element_factory_make ("queue", tmp);
g_free (tmp);
webrtc = gst_element_factory_make ("webrtcbin", peer_id);
gst_bin_add_many (GST_BIN (pipeline), q, webrtc, NULL);
srcpad = gst_element_get_static_pad (q, "src");
g_assert_nonnull (srcpad);
sinkpad = gst_element_request_pad_simple (webrtc, "sink_%u");
g_assert_nonnull (sinkpad);
ret = gst_pad_link (srcpad, sinkpad);
g_assert_cmpint (ret, ==, GST_PAD_LINK_OK);
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
tee = gst_bin_get_by_name (GST_BIN (pipeline), "audiotee");
g_assert_nonnull (tee);
srcpad = gst_element_request_pad_simple (tee, "src_%u");
g_assert_nonnull (srcpad);
gst_object_unref (tee);
sinkpad = gst_element_get_static_pad (q, "sink");
g_assert_nonnull (sinkpad);
ret = gst_pad_link (srcpad, sinkpad);
g_assert_cmpint (ret, ==, GST_PAD_LINK_OK);
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
/* This is the gstwebrtc entry point where we create the offer and so on. It
* will be called when the pipeline goes to PLAYING.
* XXX: We must connect this after webrtcbin has been linked to a source via
* get_request_pad() and before we go from NULL->READY otherwise webrtcbin
* will create an SDP offer with no media lines in it. */
if (offer)
g_signal_connect (webrtc, "on-negotiation-needed",
G_CALLBACK (on_negotiation_needed), (gpointer) peer_id);
/* We need to transmit this ICE candidate to the browser via the websockets
* signalling server. Incoming ice candidates from the browser need to be
* added by us too, see on_server_message() */
g_signal_connect (webrtc, "on-ice-candidate",
G_CALLBACK (send_ice_candidate_message), (gpointer) peer_id);
/* Incoming streams will be exposed via this signal */
g_signal_connect (webrtc, "pad-added", G_CALLBACK (on_incoming_stream),
pipeline);
/* Set to pipeline branch to PLAYING */
ret = gst_element_sync_state_with_parent (q);
g_assert_true (ret);
ret = gst_element_sync_state_with_parent (webrtc);
g_assert_true (ret);
}
static void
call_peer (const gchar * peer_id)
{
add_peer_to_pipeline (peer_id, TRUE);
}
static void
incoming_call_from_peer (const gchar * peer_id)
{
add_peer_to_pipeline (peer_id, FALSE);
}
#define STR(x) #x
#define RTP_CAPS_OPUS(x) "application/x-rtp,media=audio,encoding-name=OPUS,payload=" STR(x)
static gboolean
start_pipeline (void)
{
GstStateChangeReturn ret;
GError *error = NULL;
GstBus *bus = NULL;
/* NOTE: webrtcbin currently does not support dynamic addition/removal of
* streams, so we use a separate webrtcbin for each peer, but all of them are
* inside the same pipeline. We start by connecting it to a fakesink so that
* we can preroll early. */
pipeline = gst_parse_launch ("tee name=audiotee ! queue ! fakesink "
"audiotestsrc is-live=true wave=red-noise ! queue ! opusenc ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS (96) " ! audiotee. ", &error);
if (error) {
gst_printerr ("Failed to parse launch: %s\n", error->message);
g_error_free (error);
goto err;
}
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, bus_watch_cb, pipeline);
gst_object_unref (bus);
gst_print ("Starting pipeline, not transmitting yet\n");
ret = gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE)
goto err;
return TRUE;
err:
gst_print ("State change failure\n");
if (pipeline)
g_clear_object (&pipeline);
return FALSE;
}
static gboolean
join_room_on_server (void)
{
gchar *msg;
if (soup_websocket_connection_get_state (ws_conn) !=
SOUP_WEBSOCKET_STATE_OPEN)
return FALSE;
if (!room_id)
return FALSE;
gst_print ("Joining room %s\n", room_id);
app_state = ROOM_JOINING;
msg = g_strdup_printf ("ROOM %s", room_id);
soup_websocket_connection_send_text (ws_conn, msg);
g_free (msg);
return TRUE;
}
static gboolean
register_with_server (void)
{
gchar *hello;
if (soup_websocket_connection_get_state (ws_conn) !=
SOUP_WEBSOCKET_STATE_OPEN)
return FALSE;
gst_print ("Registering id %s with server\n", local_id);
app_state = SERVER_REGISTERING;
/* Register with the server with a random integer id. Reply will be received
* by on_server_message() */
hello = g_strdup_printf ("HELLO %s", local_id);
soup_websocket_connection_send_text (ws_conn, hello);
g_free (hello);
return TRUE;
}
static void
on_server_closed (SoupWebsocketConnection * conn G_GNUC_UNUSED,
gpointer user_data G_GNUC_UNUSED)
{
app_state = SERVER_CLOSED;
cleanup_and_quit_loop ("Server connection closed", 0);
}
static gboolean
do_registration (void)
{
if (app_state != SERVER_REGISTERING) {
cleanup_and_quit_loop ("ERROR: Received HELLO when not registering",
APP_STATE_ERROR);
return FALSE;
}
app_state = SERVER_REGISTERED;
gst_print ("Registered with server\n");
/* Ask signalling server that we want to join a room */
if (!join_room_on_server ()) {
cleanup_and_quit_loop ("ERROR: Failed to join room", ROOM_CALL_ERROR);
return FALSE;
}
return TRUE;
}
/*
* When we join a room, we are responsible for calling by starting negotiation
* with each peer in it by sending an SDP offer and ICE candidates.
*/
static void
do_join_room (const gchar * text)
{
gint ii, len;
gchar **peer_ids;
if (app_state != ROOM_JOINING) {
cleanup_and_quit_loop ("ERROR: Received ROOM_OK when not calling",
ROOM_JOIN_ERROR);
return;
}
app_state = ROOM_JOINED;
gst_print ("Room joined\n");
/* Start recording, but not transmitting */
if (!start_pipeline ()) {
cleanup_and_quit_loop ("ERROR: Failed to start pipeline", ROOM_CALL_ERROR);
return;
}
peer_ids = g_strsplit (text, " ", -1);
g_assert_cmpstr (peer_ids[0], ==, "ROOM_OK");
len = g_strv_length (peer_ids);
/* There are peers in the room already. We need to start negotiation
* (exchange SDP and ICE candidates) and transmission of media. */
if (len > 1 && strlen (peer_ids[1]) > 0) {
gst_print ("Found %i peers already in room\n", len - 1);
app_state = ROOM_CALL_OFFERING;
for (ii = 1; ii < len; ii++) {
gchar *peer_id = g_strdup (peer_ids[ii]);
gst_print ("Negotiating with peer %s\n", peer_id);
/* This might fail asynchronously */
call_peer (peer_id);
peers = g_list_prepend (peers, peer_id);
}
}
g_strfreev (peer_ids);
return;
}
static void
handle_error_message (const gchar * msg)
{
switch (app_state) {
case SERVER_CONNECTING:
app_state = SERVER_CONNECTION_ERROR;
break;
case SERVER_REGISTERING:
app_state = SERVER_REGISTRATION_ERROR;
break;
case ROOM_JOINING:
app_state = ROOM_JOIN_ERROR;
break;
case ROOM_JOINED:
case ROOM_CALL_NEGOTIATING:
case ROOM_CALL_OFFERING:
case ROOM_CALL_ANSWERING:
app_state = ROOM_CALL_ERROR;
break;
case ROOM_CALL_STARTED:
case ROOM_CALL_STOPPING:
case ROOM_CALL_STOPPED:
app_state = ROOM_CALL_ERROR;
break;
default:
app_state = APP_STATE_ERROR;
}
cleanup_and_quit_loop (msg, 0);
}
static void
on_answer_created (GstPromise * promise, const gchar * peer_id)
{
GstElement *webrtc;
GstWebRTCSessionDescription *answer;
const GstStructure *reply;
g_assert_cmpint (app_state, ==, ROOM_CALL_ANSWERING);
g_assert_cmpint (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);
promise = gst_promise_new ();
webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id);
g_assert_nonnull (webrtc);
g_signal_emit_by_name (webrtc, "set-local-description", answer, promise);
gst_promise_interrupt (promise);
gst_promise_unref (promise);
/* Send offer to peer */
send_room_peer_sdp (answer, peer_id);
gst_webrtc_session_description_free (answer);
gst_object_unref (webrtc);
app_state = ROOM_CALL_STARTED;
}
static void
handle_sdp_offer (const gchar * peer_id, const gchar * text)
{
int ret;
GstPromise *promise;
GstElement *webrtc;
GstSDPMessage *sdp;
GstWebRTCSessionDescription *offer;
g_assert_cmpint (app_state, ==, ROOM_CALL_ANSWERING);
gst_print ("Received offer:\n%s\n", text);
ret = gst_sdp_message_new (&sdp);
g_assert_cmpint (ret, ==, GST_SDP_OK);
ret = gst_sdp_message_parse_buffer ((guint8 *) text, strlen (text), sdp);
g_assert_cmpint (ret, ==, GST_SDP_OK);
offer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER, sdp);
g_assert_nonnull (offer);
/* Set remote description on our pipeline */
promise = gst_promise_new ();
webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id);
g_assert_nonnull (webrtc);
g_signal_emit_by_name (webrtc, "set-remote-description", offer, promise);
/* We don't want to be notified when the action is done */
gst_promise_interrupt (promise);
gst_promise_unref (promise);
/* Create an answer that we will send back to the peer */
promise = gst_promise_new_with_change_func (
(GstPromiseChangeFunc) on_answer_created, (gpointer) peer_id, NULL);
g_signal_emit_by_name (webrtc, "create-answer", NULL, promise);
gst_webrtc_session_description_free (offer);
gst_object_unref (webrtc);
}
static void
handle_sdp_answer (const gchar * peer_id, const gchar * text)
{
int ret;
GstPromise *promise;
GstElement *webrtc;
GstSDPMessage *sdp;
GstWebRTCSessionDescription *answer;
g_assert_cmpint (app_state, >=, ROOM_CALL_OFFERING);
gst_print ("Received answer:\n%s\n", text);
ret = gst_sdp_message_new (&sdp);
g_assert_cmpint (ret, ==, GST_SDP_OK);
ret = gst_sdp_message_parse_buffer ((guint8 *) text, strlen (text), sdp);
g_assert_cmpint (ret, ==, GST_SDP_OK);
answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER, sdp);
g_assert_nonnull (answer);
/* Set remote description on our pipeline */
promise = gst_promise_new ();
webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id);
g_assert_nonnull (webrtc);
g_signal_emit_by_name (webrtc, "set-remote-description", answer, promise);
gst_object_unref (webrtc);
/* We don't want to be notified when the action is done */
gst_promise_interrupt (promise);
gst_promise_unref (promise);
}
static gboolean
handle_peer_message (const gchar * peer_id, const gchar * msg)
{
JsonNode *root;
JsonObject *object, *child;
JsonParser *parser = json_parser_new ();
if (!json_parser_load_from_data (parser, msg, -1, NULL)) {
gst_printerr ("Unknown message '%s' from '%s', ignoring", msg, peer_id);
g_object_unref (parser);
return FALSE;
}
root = json_parser_get_root (parser);
if (!JSON_NODE_HOLDS_OBJECT (root)) {
gst_printerr ("Unknown json message '%s' from '%s', ignoring", msg,
peer_id);
g_object_unref (parser);
return FALSE;
}
gst_print ("Message from peer %s: %s\n", peer_id, msg);
object = json_node_get_object (root);
/* Check type of JSON message */
if (json_object_has_member (object, "sdp")) {
const gchar *text, *sdp_type;
g_assert_cmpint (app_state, >=, ROOM_JOINED);
child = json_object_get_object_member (object, "sdp");
if (!json_object_has_member (child, "type")) {
cleanup_and_quit_loop ("ERROR: received SDP without 'type'",
ROOM_CALL_ERROR);
return FALSE;
}
sdp_type = json_object_get_string_member (child, "type");
text = json_object_get_string_member (child, "sdp");
if (g_strcmp0 (sdp_type, "offer") == 0) {
app_state = ROOM_CALL_ANSWERING;
incoming_call_from_peer (peer_id);
handle_sdp_offer (peer_id, text);
} else if (g_strcmp0 (sdp_type, "answer") == 0) {
g_assert_cmpint (app_state, >=, ROOM_CALL_OFFERING);
handle_sdp_answer (peer_id, text);
app_state = ROOM_CALL_STARTED;
} else {
cleanup_and_quit_loop ("ERROR: invalid sdp_type", ROOM_CALL_ERROR);
return FALSE;
}
} else if (json_object_has_member (object, "ice")) {
GstElement *webrtc;
const gchar *candidate;
gint sdpmlineindex;
child = json_object_get_object_member (object, "ice");
candidate = json_object_get_string_member (child, "candidate");
sdpmlineindex = json_object_get_int_member (child, "sdpMLineIndex");
/* Add ice candidate sent by remote peer */
webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id);
g_assert_nonnull (webrtc);
g_signal_emit_by_name (webrtc, "add-ice-candidate", sdpmlineindex,
candidate);
gst_object_unref (webrtc);
} else {
gst_printerr ("Ignoring unknown JSON message:\n%s\n", msg);
}
g_object_unref (parser);
return TRUE;
}
/* One mega message handler for our asynchronous calling mechanism */
static void
on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
GBytes * message, gpointer user_data)
{
gchar *text;
switch (type) {
case SOUP_WEBSOCKET_DATA_BINARY:
gst_printerr ("Received unknown binary message, ignoring\n");
return;
case SOUP_WEBSOCKET_DATA_TEXT:{
gsize size;
const gchar *data = g_bytes_get_data (message, &size);
/* Convert to NULL-terminated string */
text = g_strndup (data, size);
break;
}
default:
g_assert_not_reached ();
}
/* Server has accepted our registration, we are ready to send commands */
if (g_strcmp0 (text, "HELLO") == 0) {
/* May fail asynchronously */
do_registration ();
/* Room-related message */
} else if (g_str_has_prefix (text, "ROOM_")) {
/* Room joined, now we can start negotiation */
if (g_str_has_prefix (text, "ROOM_OK ")) {
/* May fail asynchronously */
do_join_room (text);
} else if (g_str_has_prefix (text, "ROOM_PEER")) {
gchar **splitm = NULL;
const gchar *peer_id;
/* SDP and ICE, usually */
if (g_str_has_prefix (text, "ROOM_PEER_MSG")) {
splitm = g_strsplit (text, " ", 3);
peer_id = find_peer_from_list (splitm[1]);
g_assert_nonnull (peer_id);
/* Could be an offer or an answer, or ICE, or an arbitrary message */
handle_peer_message (peer_id, splitm[2]);
} else if (g_str_has_prefix (text, "ROOM_PEER_JOINED")) {
splitm = g_strsplit (text, " ", 2);
peers = g_list_prepend (peers, g_strdup (splitm[1]));
peer_id = find_peer_from_list (splitm[1]);
g_assert_nonnull (peer_id);
gst_print ("Peer %s has joined the room\n", peer_id);
} else if (g_str_has_prefix (text, "ROOM_PEER_LEFT")) {
splitm = g_strsplit (text, " ", 2);
peer_id = find_peer_from_list (splitm[1]);
g_assert_nonnull (peer_id);
peers = g_list_remove (peers, peer_id);
gst_print ("Peer %s has left the room\n", peer_id);
remove_peer_from_pipeline (peer_id);
g_free ((gchar *) peer_id);
/* TODO: cleanup pipeline */
} else {
gst_printerr ("WARNING: Ignoring unknown message %s\n", text);
}
g_strfreev (splitm);
} else {
goto err;
}
/* Handle errors */
} else if (g_str_has_prefix (text, "ERROR")) {
handle_error_message (text);
} else {
goto err;
}
out:
g_free (text);
return;
err:
{
gchar *err_s = g_strdup_printf ("ERROR: unknown message %s", text);
cleanup_and_quit_loop (err_s, 0);
g_free (err_s);
goto out;
}
}
static void
on_server_connected (SoupSession * session, GAsyncResult * res,
SoupMessage * msg)
{
GError *error = NULL;
ws_conn = soup_session_websocket_connect_finish (session, res, &error);
if (error) {
cleanup_and_quit_loop (error->message, SERVER_CONNECTION_ERROR);
g_error_free (error);
return;
}
g_assert_nonnull (ws_conn);
app_state = SERVER_CONNECTED;
gst_print ("Connected to signalling server\n");
g_signal_connect (ws_conn, "closed", G_CALLBACK (on_server_closed), NULL);
g_signal_connect (ws_conn, "message", G_CALLBACK (on_server_message), NULL);
/* Register with the server so it knows about us and can accept commands
* responses from the server will be handled in on_server_message() above */
register_with_server ();
}
/*
* Connect to the signalling server. This is the entrypoint for everything else.
*/
static void
connect_to_websocket_server_async (void)
{
SoupLogger *logger;
SoupMessage *message;
SoupSession *session;
const char *https_aliases[] = { "wss", NULL };
session = soup_session_new_with_options (SOUP_SESSION_SSL_STRICT, strict_ssl,
SOUP_SESSION_SSL_USE_SYSTEM_CA_FILE, TRUE,
//SOUP_SESSION_SSL_CA_FILE, "/etc/ssl/certs/ca-bundle.crt",
SOUP_SESSION_HTTPS_ALIASES, https_aliases, NULL);
logger = soup_logger_new (SOUP_LOGGER_LOG_BODY, -1);
soup_session_add_feature (session, SOUP_SESSION_FEATURE (logger));
g_object_unref (logger);
message = soup_message_new (SOUP_METHOD_GET, server_url);
gst_print ("Connecting to server...\n");
/* Once connected, we will register */
soup_session_websocket_connect_async (session, message, NULL, NULL, NULL,
(GAsyncReadyCallback) on_server_connected, message);
app_state = SERVER_CONNECTING;
}
static gboolean
check_plugins (void)
{
int i;
gboolean ret;
GstRegistry *registry;
const gchar *needed[] = { "opus", "nice", "webrtc", "dtls", "srtp",
"rtpmanager", "audiotestsrc", NULL
};
registry = gst_registry_get ();
ret = TRUE;
for (i = 0; i < g_strv_length ((gchar **) needed); i++) {
GstPlugin *plugin;
plugin = gst_registry_find_plugin (registry, needed[i]);
if (!plugin) {
gst_print ("Required gstreamer plugin '%s' not found\n", needed[i]);
ret = FALSE;
continue;
}
gst_object_unref (plugin);
}
return ret;
}
int
main (int argc, char *argv[])
{
GOptionContext *context;
GstBus *bus;
GError *error = NULL;
context = g_option_context_new ("- gstreamer webrtc sendrecv demo");
g_option_context_add_main_entries (context, entries, NULL);
g_option_context_add_group (context, gst_init_get_option_group ());
if (!g_option_context_parse (context, &argc, &argv, &error)) {
gst_printerr ("Error initializing: %s\n", error->message);
return -1;
}
if (!check_plugins ())
return -1;
if (!room_id) {
gst_printerr ("--room-id is a required argument\n");
return -1;
}
if (!local_id)
local_id = g_strdup_printf ("%s-%i", g_get_user_name (),
g_random_int_range (10, 10000));
/* Sanitize by removing whitespace, modifies string in-place */
g_strdelimit (local_id, " \t\n\r", '-');
gst_print ("Our local id is %s\n", local_id);
if (!server_url)
server_url = g_strdup (default_server_url);
/* Don't use strict ssl when running a localhost server, because
* it's probably a test server with a self-signed certificate */
{
GstUri *uri = gst_uri_from_string (server_url);
if (g_strcmp0 ("localhost", gst_uri_get_host (uri)) == 0 ||
g_strcmp0 ("127.0.0.1", gst_uri_get_host (uri)) == 0)
strict_ssl = FALSE;
gst_uri_unref (uri);
}
loop = g_main_loop_new (NULL, FALSE);
connect_to_websocket_server_async ();
g_main_loop_run (loop);
gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_NULL);
gst_print ("Pipeline stopped\n");
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_remove_watch (bus);
gst_object_unref (bus);
gst_object_unref (pipeline);
g_free (server_url);
g_free (local_id);
g_free (room_id);
return 0;
}