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gstreamer/subprojects/gst-examples/webrtc/sendonly/webrtc-recvonly-h264.c

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#include <locale.h>
#include <glib.h>
#include <gst/gst.h>
#include <gst/sdp/sdp.h>
#ifdef G_OS_UNIX
#include <glib-unix.h>
#endif
#define GST_USE_UNSTABLE_API
#include <gst/webrtc/webrtc.h>
#include <libsoup/soup.h>
#include <json-glib/json-glib.h>
#include <string.h>
/* This example is a standalone app which serves a web page
* and configures webrtcbin to receive an H.264 video feed, and to
* send+recv an Opus audio stream */
#define RTP_PAYLOAD_TYPE "96"
#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
#define SOUP_HTTP_PORT 57778
#define STUN_SERVER "stun.l.google.com:19302"
typedef struct _ReceiverEntry ReceiverEntry;
ReceiverEntry *create_receiver_entry (SoupWebsocketConnection * connection);
void destroy_receiver_entry (gpointer receiver_entry_ptr);
GstPadProbeReturn payloader_caps_event_probe_cb (GstPad * pad,
GstPadProbeInfo * info, gpointer user_data);
void on_offer_created_cb (GstPromise * promise, gpointer user_data);
void on_negotiation_needed_cb (GstElement * webrtcbin, gpointer user_data);
void on_ice_candidate_cb (GstElement * webrtcbin, guint mline_index,
gchar * candidate, gpointer user_data);
void soup_websocket_message_cb (SoupWebsocketConnection * connection,
SoupWebsocketDataType data_type, GBytes * message, gpointer user_data);
void soup_websocket_closed_cb (SoupWebsocketConnection * connection,
gpointer user_data);
void soup_http_handler (SoupServer * soup_server, SoupMessage * message,
const char *path, GHashTable * query, SoupClientContext * client_context,
gpointer user_data);
void soup_websocket_handler (G_GNUC_UNUSED SoupServer * server,
SoupWebsocketConnection * connection, const char *path,
SoupClientContext * client_context, gpointer user_data);
static gchar *get_string_from_json_object (JsonObject * object);
struct _ReceiverEntry
{
SoupWebsocketConnection *connection;
GstElement *pipeline;
GstElement *webrtcbin;
};
const gchar *html_source = " \n \
<html> \n \
<head> \n \
<script type=\"text/javascript\" src=\"https://webrtc.github.io/adapter/adapter-latest.js\"></script> \n \
<script type=\"text/javascript\"> \n \
var html5VideoElement; \n \
var websocketConnection; \n \
var webrtcPeerConnection; \n \
var webrtcConfiguration; \n \
var reportError; \n \
\n \
function getLocalStream() { \n \
var constraints = {\"video\":true,\"audio\":true}; \n \
if (navigator.mediaDevices.getUserMedia) { \n \
return navigator.mediaDevices.getUserMedia(constraints); \n \
} \n \
} \n \
\n \
function onLocalDescription(desc) { \n \
console.log(\"Local description: \" + JSON.stringify(desc)); \n \
webrtcPeerConnection.setLocalDescription(desc).then(function() { \n \
websocketConnection.send(JSON.stringify({ type: \"sdp\", \"data\": webrtcPeerConnection.localDescription })); \n \
}).catch(reportError); \n \
} \n \
\n \
\n \
function onIncomingSDP(sdp) { \n \
console.log(\"Incoming SDP: \" + JSON.stringify(sdp)); \n \
webrtcPeerConnection.setRemoteDescription(sdp).catch(reportError); \n \
/* Send our video/audio to the other peer */ \n \
local_stream_promise = getLocalStream().then((stream) => { \n \
console.log('Adding local stream'); \n \
webrtcPeerConnection.addStream(stream); \n \
webrtcPeerConnection.createAnswer().then(onLocalDescription).catch(reportError); \n \
}); \n \
} \n \
\n \
\n \
function onIncomingICE(ice) { \n \
var candidate = new RTCIceCandidate(ice); \n \
console.log(\"Incoming ICE: \" + JSON.stringify(ice)); \n \
webrtcPeerConnection.addIceCandidate(candidate).catch(reportError); \n \
} \n \
\n \
\n \
function onAddRemoteStream(event) { \n \
html5VideoElement.srcObject = event.streams[0]; \n \
} \n \
\n \
\n \
function onIceCandidate(event) { \n \
if (event.candidate == null) \n \
return; \n \
\n \
console.log(\"Sending ICE candidate out: \" + JSON.stringify(event.candidate)); \n \
websocketConnection.send(JSON.stringify({ \"type\": \"ice\", \"data\": event.candidate })); \n \
} \n \
\n \
\n \
function onServerMessage(event) { \n \
var msg; \n \
\n \
try { \n \
msg = JSON.parse(event.data); \n \
} catch (e) { \n \
return; \n \
} \n \
\n \
if (!webrtcPeerConnection) { \n \
webrtcPeerConnection = new RTCPeerConnection(webrtcConfiguration); \n \
webrtcPeerConnection.ontrack = onAddRemoteStream; \n \
webrtcPeerConnection.onicecandidate = onIceCandidate; \n \
} \n \
\n \
switch (msg.type) { \n \
case \"sdp\": onIncomingSDP(msg.data); break; \n \
case \"ice\": onIncomingICE(msg.data); break; \n \
default: break; \n \
} \n \
} \n \
\n \
\n \
function playStream(videoElement, hostname, port, path, configuration, reportErrorCB) { \n \
var l = window.location;\n \
var wsHost = (hostname != undefined) ? hostname : l.hostname; \n \
var wsPort = (port != undefined) ? port : l.port; \n \
var wsPath = (path != undefined) ? path : \"ws\"; \n \
if (wsPort) \n\
wsPort = \":\" + wsPort; \n\
var wsUrl = \"ws://\" + wsHost + wsPort + \"/\" + wsPath; \n \
\n \
html5VideoElement = videoElement; \n \
webrtcConfiguration = configuration; \n \
reportError = (reportErrorCB != undefined) ? reportErrorCB : function(text) {}; \n \
\n \
websocketConnection = new WebSocket(wsUrl); \n \
websocketConnection.addEventListener(\"message\", onServerMessage); \n \
} \n \
\n \
window.onload = function() { \n \
var vidstream = document.getElementById(\"stream\"); \n \
var config = { 'iceServers': [{ 'urls': 'stun:" STUN_SERVER "' }] }; \n\
playStream(vidstream, null, null, null, config, function (errmsg) { console.error(errmsg); }); \n \
}; \n \
\n \
</script> \n \
</head> \n \
\n \
<body> \n \
<div> \n \
<video id=\"stream\" autoplay playsinline>Your browser does not support video</video> \n \
</div> \n \
</body> \n \
</html> \n \
";
static void
handle_media_stream (GstPad * pad, GstElement * pipe, const char *convert_name,
const char *sink_name)
{
GstPad *qpad;
GstElement *q, *conv, *resample, *sink;
GstPadLinkReturn ret;
gst_print ("Trying to handle stream with %s ! %s", convert_name, sink_name);
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);
if (g_strcmp0 (convert_name, "audioconvert") == 0) {
/* Might also need to resample, so add it just in case.
* Will be a no-op if it's not required. */
resample = gst_element_factory_make ("audioresample", NULL);
g_assert_nonnull (resample);
gst_bin_add_many (GST_BIN (pipe), q, conv, resample, sink, NULL);
gst_element_sync_state_with_parent (q);
gst_element_sync_state_with_parent (conv);
gst_element_sync_state_with_parent (resample);
gst_element_sync_state_with_parent (sink);
gst_element_link_many (q, conv, resample, sink, NULL);
} else {
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_cmphex (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,
ReceiverEntry * receiver_entry)
{
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), receiver_entry->pipeline);
gst_bin_add (GST_BIN (receiver_entry->pipeline), 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 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;
}
ReceiverEntry *
create_receiver_entry (SoupWebsocketConnection * connection)
{
GError *error;
ReceiverEntry *receiver_entry;
GstCaps *video_caps;
GstWebRTCRTPTransceiver *trans = NULL;
GstBus *bus;
receiver_entry = g_new0 (ReceiverEntry, 1);
receiver_entry->connection = connection;
g_object_ref (G_OBJECT (connection));
g_signal_connect (G_OBJECT (connection), "message",
G_CALLBACK (soup_websocket_message_cb), (gpointer) receiver_entry);
error = NULL;
receiver_entry->pipeline =
gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://"
STUN_SERVER " "
"audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS "97 ! webrtcbin. ", &error);
if (error != NULL) {
g_error ("Could not create WebRTC pipeline: %s\n", error->message);
g_error_free (error);
goto cleanup;
}
receiver_entry->webrtcbin =
gst_bin_get_by_name (GST_BIN (receiver_entry->pipeline), "webrtcbin");
g_assert (receiver_entry->webrtcbin != NULL);
/* Incoming streams will be exposed via this signal */
g_signal_connect (receiver_entry->webrtcbin, "pad-added",
G_CALLBACK (on_incoming_stream), receiver_entry);
#if 0
GstElement *rtpbin =
gst_bin_get_by_name (GST_BIN (receiver_entry->webrtcbin), "rtpbin");
g_object_set (rtpbin, "latency", 40, NULL);
gst_object_unref (rtpbin);
#endif
// Create a 2nd transceiver for the receive only video stream
video_caps =
gst_caps_from_string
("application/x-rtp,media=video,encoding-name=H264,payload="
RTP_PAYLOAD_TYPE
",clock-rate=90000,packetization-mode=(string)1, profile-level-id=(string)42c016");
g_signal_emit_by_name (receiver_entry->webrtcbin, "add-transceiver",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, video_caps, &trans);
gst_caps_unref (video_caps);
gst_object_unref (trans);
g_signal_connect (receiver_entry->webrtcbin, "on-negotiation-needed",
G_CALLBACK (on_negotiation_needed_cb), (gpointer) receiver_entry);
g_signal_connect (receiver_entry->webrtcbin, "on-ice-candidate",
G_CALLBACK (on_ice_candidate_cb), (gpointer) receiver_entry);
bus = gst_pipeline_get_bus (GST_PIPELINE (receiver_entry->pipeline));
gst_bus_add_watch (bus, bus_watch_cb, receiver_entry->pipeline);
gst_object_unref (bus);
if (gst_element_set_state (receiver_entry->pipeline, GST_STATE_PLAYING) ==
GST_STATE_CHANGE_FAILURE)
g_error ("Error starting pipeline");
return receiver_entry;
cleanup:
destroy_receiver_entry ((gpointer) receiver_entry);
return NULL;
}
void
destroy_receiver_entry (gpointer receiver_entry_ptr)
{
ReceiverEntry *receiver_entry = (ReceiverEntry *) receiver_entry_ptr;
g_assert (receiver_entry != NULL);
if (receiver_entry->pipeline != NULL) {
GstBus *bus;
gst_element_set_state (GST_ELEMENT (receiver_entry->pipeline),
GST_STATE_NULL);
bus = gst_pipeline_get_bus (GST_PIPELINE (receiver_entry->pipeline));
gst_bus_remove_watch (bus);
gst_object_unref (bus);
gst_object_unref (GST_OBJECT (receiver_entry->webrtcbin));
gst_object_unref (GST_OBJECT (receiver_entry->pipeline));
}
if (receiver_entry->connection != NULL)
g_object_unref (G_OBJECT (receiver_entry->connection));
g_free (receiver_entry);
}
void
on_offer_created_cb (GstPromise * promise, gpointer user_data)
{
gchar *sdp_string;
gchar *json_string;
JsonObject *sdp_json;
JsonObject *sdp_data_json;
GstStructure const *reply;
GstPromise *local_desc_promise;
GstWebRTCSessionDescription *offer = NULL;
ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
&offer, NULL);
gst_promise_unref (promise);
local_desc_promise = gst_promise_new ();
g_signal_emit_by_name (receiver_entry->webrtcbin, "set-local-description",
offer, local_desc_promise);
gst_promise_interrupt (local_desc_promise);
gst_promise_unref (local_desc_promise);
sdp_string = gst_sdp_message_as_text (offer->sdp);
gst_print ("Negotiation offer created:\n%s\n", sdp_string);
sdp_json = json_object_new ();
json_object_set_string_member (sdp_json, "type", "sdp");
sdp_data_json = json_object_new ();
json_object_set_string_member (sdp_data_json, "type", "offer");
json_object_set_string_member (sdp_data_json, "sdp", sdp_string);
json_object_set_object_member (sdp_json, "data", sdp_data_json);
json_string = get_string_from_json_object (sdp_json);
json_object_unref (sdp_json);
soup_websocket_connection_send_text (receiver_entry->connection, json_string);
g_free (json_string);
g_free (sdp_string);
gst_webrtc_session_description_free (offer);
}
void
on_negotiation_needed_cb (GstElement * webrtcbin, gpointer user_data)
{
GstPromise *promise;
ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;
gst_print ("Creating negotiation offer\n");
promise = gst_promise_new_with_change_func (on_offer_created_cb,
(gpointer) receiver_entry, NULL);
g_signal_emit_by_name (G_OBJECT (webrtcbin), "create-offer", NULL, promise);
}
void
on_ice_candidate_cb (G_GNUC_UNUSED GstElement * webrtcbin, guint mline_index,
gchar * candidate, gpointer user_data)
{
JsonObject *ice_json;
JsonObject *ice_data_json;
gchar *json_string;
ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;
ice_json = json_object_new ();
json_object_set_string_member (ice_json, "type", "ice");
ice_data_json = json_object_new ();
json_object_set_int_member (ice_data_json, "sdpMLineIndex", mline_index);
json_object_set_string_member (ice_data_json, "candidate", candidate);
json_object_set_object_member (ice_json, "data", ice_data_json);
json_string = get_string_from_json_object (ice_json);
json_object_unref (ice_json);
soup_websocket_connection_send_text (receiver_entry->connection, json_string);
g_free (json_string);
}
void
soup_websocket_message_cb (G_GNUC_UNUSED SoupWebsocketConnection * connection,
SoupWebsocketDataType data_type, GBytes * message, gpointer user_data)
{
gsize size;
const gchar *data;
gchar *data_string;
const gchar *type_string;
JsonNode *root_json;
JsonObject *root_json_object;
JsonObject *data_json_object;
JsonParser *json_parser = NULL;
ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;
switch (data_type) {
case SOUP_WEBSOCKET_DATA_BINARY:
g_error ("Received unknown binary message, ignoring\n");
return;
case SOUP_WEBSOCKET_DATA_TEXT:
data = g_bytes_get_data (message, &size);
/* Convert to NULL-terminated string */
data_string = g_strndup (data, size);
break;
default:
g_assert_not_reached ();
}
json_parser = json_parser_new ();
if (!json_parser_load_from_data (json_parser, data_string, -1, NULL))
goto unknown_message;
root_json = json_parser_get_root (json_parser);
if (!JSON_NODE_HOLDS_OBJECT (root_json))
goto unknown_message;
root_json_object = json_node_get_object (root_json);
if (!json_object_has_member (root_json_object, "type")) {
g_error ("Received message without type field\n");
goto cleanup;
}
type_string = json_object_get_string_member (root_json_object, "type");
if (!json_object_has_member (root_json_object, "data")) {
g_error ("Received message without data field\n");
goto cleanup;
}
data_json_object = json_object_get_object_member (root_json_object, "data");
if (g_strcmp0 (type_string, "sdp") == 0) {
const gchar *sdp_type_string;
const gchar *sdp_string;
GstPromise *promise;
GstSDPMessage *sdp;
GstWebRTCSessionDescription *answer;
int ret;
if (!json_object_has_member (data_json_object, "type")) {
g_error ("Received SDP message without type field\n");
goto cleanup;
}
sdp_type_string = json_object_get_string_member (data_json_object, "type");
if (g_strcmp0 (sdp_type_string, "answer") != 0) {
g_error ("Expected SDP message type \"answer\", got \"%s\"\n",
sdp_type_string);
goto cleanup;
}
if (!json_object_has_member (data_json_object, "sdp")) {
g_error ("Received SDP message without SDP string\n");
goto cleanup;
}
sdp_string = json_object_get_string_member (data_json_object, "sdp");
gst_print ("Received SDP:\n%s\n", sdp_string);
ret = gst_sdp_message_new (&sdp);
g_assert_cmphex (ret, ==, GST_SDP_OK);
ret =
gst_sdp_message_parse_buffer ((guint8 *) sdp_string,
strlen (sdp_string), sdp);
if (ret != GST_SDP_OK) {
g_error ("Could not parse SDP string\n");
goto cleanup;
}
answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER,
sdp);
g_assert_nonnull (answer);
promise = gst_promise_new ();
g_signal_emit_by_name (receiver_entry->webrtcbin, "set-remote-description",
answer, promise);
gst_promise_interrupt (promise);
gst_promise_unref (promise);
gst_webrtc_session_description_free (answer);
} else if (g_strcmp0 (type_string, "ice") == 0) {
guint mline_index;
const gchar *candidate_string;
if (!json_object_has_member (data_json_object, "sdpMLineIndex")) {
g_error ("Received ICE message without mline index\n");
goto cleanup;
}
mline_index =
json_object_get_int_member (data_json_object, "sdpMLineIndex");
if (!json_object_has_member (data_json_object, "candidate")) {
g_error ("Received ICE message without ICE candidate string\n");
goto cleanup;
}
candidate_string = json_object_get_string_member (data_json_object,
"candidate");
gst_print ("Received ICE candidate with mline index %u; candidate: %s\n",
mline_index, candidate_string);
g_signal_emit_by_name (receiver_entry->webrtcbin, "add-ice-candidate",
mline_index, candidate_string);
} else
goto unknown_message;
cleanup:
if (json_parser != NULL)
g_object_unref (G_OBJECT (json_parser));
g_free (data_string);
return;
unknown_message:
g_error ("Unknown message \"%s\", ignoring", data_string);
goto cleanup;
}
void
soup_websocket_closed_cb (SoupWebsocketConnection * connection,
gpointer user_data)
{
GHashTable *receiver_entry_table = (GHashTable *) user_data;
g_hash_table_remove (receiver_entry_table, connection);
gst_print ("Closed websocket connection %p\n", (gpointer) connection);
}
void
soup_http_handler (G_GNUC_UNUSED SoupServer * soup_server,
SoupMessage * message, const char *path, G_GNUC_UNUSED GHashTable * query,
G_GNUC_UNUSED SoupClientContext * client_context,
G_GNUC_UNUSED gpointer user_data)
{
SoupBuffer *soup_buffer;
if ((g_strcmp0 (path, "/") != 0) && (g_strcmp0 (path, "/index.html") != 0)) {
soup_message_set_status (message, SOUP_STATUS_NOT_FOUND);
return;
}
soup_buffer =
soup_buffer_new (SOUP_MEMORY_STATIC, html_source, strlen (html_source));
soup_message_headers_set_content_type (message->response_headers, "text/html",
NULL);
soup_message_body_append_buffer (message->response_body, soup_buffer);
soup_buffer_free (soup_buffer);
soup_message_set_status (message, SOUP_STATUS_OK);
}
void
soup_websocket_handler (G_GNUC_UNUSED SoupServer * server,
SoupWebsocketConnection * connection, G_GNUC_UNUSED const char *path,
G_GNUC_UNUSED SoupClientContext * client_context, gpointer user_data)
{
ReceiverEntry *receiver_entry;
GHashTable *receiver_entry_table = (GHashTable *) user_data;
gst_print ("Processing new websocket connection %p", (gpointer) connection);
g_signal_connect (G_OBJECT (connection), "closed",
G_CALLBACK (soup_websocket_closed_cb), (gpointer) receiver_entry_table);
receiver_entry = create_receiver_entry (connection);
g_hash_table_replace (receiver_entry_table, connection, receiver_entry);
}
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;
}
#ifdef G_OS_UNIX
gboolean
exit_sighandler (gpointer user_data)
{
gst_print ("Caught signal, stopping mainloop\n");
GMainLoop *mainloop = (GMainLoop *) user_data;
g_main_loop_quit (mainloop);
return TRUE;
}
#endif
int
main (int argc, char *argv[])
{
GMainLoop *mainloop;
SoupServer *soup_server;
GHashTable *receiver_entry_table;
setlocale (LC_ALL, "");
gst_init (&argc, &argv);
receiver_entry_table =
g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL,
destroy_receiver_entry);
mainloop = g_main_loop_new (NULL, FALSE);
g_assert (mainloop != NULL);
#ifdef G_OS_UNIX
g_unix_signal_add (SIGINT, exit_sighandler, mainloop);
g_unix_signal_add (SIGTERM, exit_sighandler, mainloop);
#endif
soup_server =
soup_server_new (SOUP_SERVER_SERVER_HEADER, "webrtc-soup-server", NULL);
soup_server_add_handler (soup_server, "/", soup_http_handler, NULL, NULL);
soup_server_add_websocket_handler (soup_server, "/ws", NULL, NULL,
soup_websocket_handler, (gpointer) receiver_entry_table, NULL);
soup_server_listen_all (soup_server, SOUP_HTTP_PORT,
(SoupServerListenOptions) 0, NULL);
gst_print ("WebRTC page link: http://127.0.0.1:%d/\n", (gint) SOUP_HTTP_PORT);
g_main_loop_run (mainloop);
g_object_unref (G_OBJECT (soup_server));
g_hash_table_destroy (receiver_entry_table);
g_main_loop_unref (mainloop);
gst_deinit ();
return 0;
}
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