mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-13 03:46:34 +00:00
b7c0e8bc84
Using MULTIOPUS breaks with most browsers Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3675>
715 lines
22 KiB
C
715 lines
22 KiB
C
#include <locale.h>
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#include <glib.h>
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#include <gst/gst.h>
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#include <gst/sdp/sdp.h>
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#ifdef G_OS_UNIX
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#include <glib-unix.h>
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#endif
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#define GST_USE_UNSTABLE_API
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#include <gst/webrtc/webrtc.h>
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#include <libsoup/soup.h>
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#include <json-glib/json-glib.h>
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#include <string.h>
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#define RTP_PAYLOAD_TYPE "96"
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#define RTP_AUDIO_PAYLOAD_TYPE "97"
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#define SOUP_HTTP_PORT 57778
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#define STUN_SERVER "stun.l.google.com:19302"
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#ifdef G_OS_WIN32
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#define VIDEO_SRC "mfvideosrc"
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#else
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#define VIDEO_SRC "v4l2src"
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#endif
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gchar *video_priority = NULL;
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gchar *audio_priority = NULL;
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typedef struct _ReceiverEntry ReceiverEntry;
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ReceiverEntry *create_receiver_entry (SoupWebsocketConnection * connection);
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void destroy_receiver_entry (gpointer receiver_entry_ptr);
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void on_offer_created_cb (GstPromise * promise, gpointer user_data);
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void on_negotiation_needed_cb (GstElement * webrtcbin, gpointer user_data);
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void on_ice_candidate_cb (GstElement * webrtcbin, guint mline_index,
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gchar * candidate, gpointer user_data);
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void soup_websocket_message_cb (SoupWebsocketConnection * connection,
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SoupWebsocketDataType data_type, GBytes * message, gpointer user_data);
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void soup_websocket_closed_cb (SoupWebsocketConnection * connection,
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gpointer user_data);
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void soup_http_handler (SoupServer * soup_server, SoupMessage * message,
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const char *path, GHashTable * query, SoupClientContext * client_context,
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gpointer user_data);
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void soup_websocket_handler (G_GNUC_UNUSED SoupServer * server,
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SoupWebsocketConnection * connection, const char *path,
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SoupClientContext * client_context, gpointer user_data);
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static gchar *get_string_from_json_object (JsonObject * object);
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struct _ReceiverEntry
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{
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SoupWebsocketConnection *connection;
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GstElement *pipeline;
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GstElement *webrtcbin;
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};
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const gchar *html_source = " \n \
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<html> \n \
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<head> \n \
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<script type=\"text/javascript\" src=\"https://webrtc.github.io/adapter/adapter-latest.js\"></script> \n \
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<script type=\"text/javascript\"> \n \
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var html5VideoElement; \n \
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var websocketConnection; \n \
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var webrtcPeerConnection; \n \
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var webrtcConfiguration; \n \
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var reportError; \n \
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\n \
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\n \
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function onLocalDescription(desc) { \n \
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console.log(\"Local description: \" + JSON.stringify(desc)); \n \
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webrtcPeerConnection.setLocalDescription(desc).then(function() { \n \
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websocketConnection.send(JSON.stringify({ type: \"sdp\", \"data\": webrtcPeerConnection.localDescription })); \n \
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}).catch(reportError); \n \
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} \n \
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\n \
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\n \
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function onIncomingSDP(sdp) { \n \
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console.log(\"Incoming SDP: \" + JSON.stringify(sdp)); \n \
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webrtcPeerConnection.setRemoteDescription(sdp).catch(reportError); \n \
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webrtcPeerConnection.createAnswer().then(onLocalDescription).catch(reportError); \n \
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} \n \
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\n \
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\n \
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function onIncomingICE(ice) { \n \
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var candidate = new RTCIceCandidate(ice); \n \
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console.log(\"Incoming ICE: \" + JSON.stringify(ice)); \n \
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webrtcPeerConnection.addIceCandidate(candidate).catch(reportError); \n \
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} \n \
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\n \
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\n \
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function onAddRemoteStream(event) { \n \
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html5VideoElement.srcObject = event.streams[0]; \n \
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} \n \
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\n \
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\n \
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function onIceCandidate(event) { \n \
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if (event.candidate == null) \n \
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return; \n \
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\n \
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console.log(\"Sending ICE candidate out: \" + JSON.stringify(event.candidate)); \n \
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websocketConnection.send(JSON.stringify({ \"type\": \"ice\", \"data\": event.candidate })); \n \
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} \n \
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\n \
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\n \
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function onServerMessage(event) { \n \
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var msg; \n \
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\n \
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try { \n \
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msg = JSON.parse(event.data); \n \
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} catch (e) { \n \
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return; \n \
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} \n \
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\n \
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if (!webrtcPeerConnection) { \n \
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webrtcPeerConnection = new RTCPeerConnection(webrtcConfiguration); \n \
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webrtcPeerConnection.ontrack = onAddRemoteStream; \n \
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webrtcPeerConnection.onicecandidate = onIceCandidate; \n \
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} \n \
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\n \
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switch (msg.type) { \n \
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case \"sdp\": onIncomingSDP(msg.data); break; \n \
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case \"ice\": onIncomingICE(msg.data); break; \n \
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default: break; \n \
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} \n \
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} \n \
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\n \
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\n \
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function playStream(videoElement, hostname, port, path, configuration, reportErrorCB) { \n \
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var l = window.location;\n \
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var wsHost = (hostname != undefined) ? hostname : l.hostname; \n \
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var wsPort = (port != undefined) ? port : l.port; \n \
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var wsPath = (path != undefined) ? path : \"ws\"; \n \
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if (wsPort) \n\
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wsPort = \":\" + wsPort; \n\
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var wsUrl = \"ws://\" + wsHost + wsPort + \"/\" + wsPath; \n \
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\n \
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html5VideoElement = videoElement; \n \
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webrtcConfiguration = configuration; \n \
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reportError = (reportErrorCB != undefined) ? reportErrorCB : function(text) {}; \n \
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\n \
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websocketConnection = new WebSocket(wsUrl); \n \
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websocketConnection.addEventListener(\"message\", onServerMessage); \n \
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} \n \
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\n \
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window.onload = function() { \n \
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var vidstream = document.getElementById(\"stream\"); \n \
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var config = { 'iceServers': [{ 'urls': 'stun:" STUN_SERVER "' }] }; \n\
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playStream(vidstream, null, null, null, config, function (errmsg) { console.error(errmsg); }); \n \
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}; \n \
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\n \
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</script> \n \
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</head> \n \
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\n \
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<body> \n \
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<div> \n \
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<video id=\"stream\" autoplay playsinline>Your browser does not support video</video> \n \
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</div> \n \
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</body> \n \
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</html> \n \
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";
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static gboolean
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bus_watch_cb (GstBus * bus, GstMessage * message, gpointer user_data)
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{
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GstPipeline *pipeline = user_data;
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switch (GST_MESSAGE_TYPE (message)) {
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case GST_MESSAGE_ERROR:
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{
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GError *error = NULL;
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gchar *debug = NULL;
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gst_message_parse_error (message, &error, &debug);
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g_error ("Error on bus: %s (debug: %s)", error->message, debug);
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g_error_free (error);
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g_free (debug);
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break;
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}
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case GST_MESSAGE_WARNING:
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{
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GError *error = NULL;
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gchar *debug = NULL;
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gst_message_parse_warning (message, &error, &debug);
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g_warning ("Warning on bus: %s (debug: %s)", error->message, debug);
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g_error_free (error);
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g_free (debug);
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break;
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}
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case GST_MESSAGE_LATENCY:
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gst_bin_recalculate_latency (GST_BIN (pipeline));
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break;
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default:
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break;
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}
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return G_SOURCE_CONTINUE;
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}
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static GstWebRTCPriorityType
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_priority_from_string (const gchar * s)
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{
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GEnumClass *klass =
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(GEnumClass *) g_type_class_ref (GST_TYPE_WEBRTC_PRIORITY_TYPE);
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GEnumValue *en;
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g_return_val_if_fail (klass, 0);
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if (!(en = g_enum_get_value_by_name (klass, s)))
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en = g_enum_get_value_by_nick (klass, s);
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g_type_class_unref (klass);
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if (en)
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return en->value;
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return 0;
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}
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ReceiverEntry *
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create_receiver_entry (SoupWebsocketConnection * connection)
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{
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GError *error;
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ReceiverEntry *receiver_entry;
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GstWebRTCRTPTransceiver *trans;
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GArray *transceivers;
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GstBus *bus;
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receiver_entry = g_slice_alloc0 (sizeof (ReceiverEntry));
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receiver_entry->connection = connection;
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g_object_ref (G_OBJECT (connection));
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g_signal_connect (G_OBJECT (connection), "message",
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G_CALLBACK (soup_websocket_message_cb), (gpointer) receiver_entry);
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error = NULL;
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receiver_entry->pipeline =
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gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://"
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STUN_SERVER " "
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VIDEO_SRC
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" ! videorate ! videoscale ! video/x-raw,width=640,height=360,framerate=15/1 ! videoconvert ! queue max-size-buffers=1 ! x264enc bitrate=600 speed-preset=ultrafast tune=zerolatency key-int-max=15 ! video/x-h264,profile=constrained-baseline ! queue max-size-time=100000000 ! h264parse ! "
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"rtph264pay config-interval=-1 name=payloader aggregate-mode=zero-latency ! "
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"application/x-rtp,media=video,encoding-name=H264,payload="
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RTP_PAYLOAD_TYPE " ! webrtcbin. "
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"autoaudiosrc ! queue max-size-buffers=1 leaky=downstream"
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" ! audioconvert ! audioresample ! opusenc ! rtpopuspay pt="
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RTP_AUDIO_PAYLOAD_TYPE " ! application/x-rtp, encoding-name=OPUS !"
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" webrtcbin. ", &error);
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if (error != NULL) {
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g_error ("Could not create WebRTC pipeline: %s\n", error->message);
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g_error_free (error);
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goto cleanup;
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}
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receiver_entry->webrtcbin =
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gst_bin_get_by_name (GST_BIN (receiver_entry->pipeline), "webrtcbin");
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g_assert (receiver_entry->webrtcbin != NULL);
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g_signal_emit_by_name (receiver_entry->webrtcbin, "get-transceivers",
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&transceivers);
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g_assert (transceivers != NULL && transceivers->len > 1);
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trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);
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g_object_set (trans, "direction",
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, NULL);
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if (video_priority) {
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GstWebRTCPriorityType priority;
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priority = _priority_from_string (video_priority);
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if (priority) {
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GstWebRTCRTPSender *sender;
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g_object_get (trans, "sender", &sender, NULL);
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gst_webrtc_rtp_sender_set_priority (sender, priority);
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g_object_unref (sender);
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}
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}
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trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
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g_object_set (trans, "direction",
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, NULL);
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if (audio_priority) {
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GstWebRTCPriorityType priority;
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priority = _priority_from_string (audio_priority);
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if (priority) {
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GstWebRTCRTPSender *sender;
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g_object_get (trans, "sender", &sender, NULL);
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gst_webrtc_rtp_sender_set_priority (sender, priority);
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g_object_unref (sender);
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}
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}
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g_array_unref (transceivers);
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g_signal_connect (receiver_entry->webrtcbin, "on-negotiation-needed",
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G_CALLBACK (on_negotiation_needed_cb), (gpointer) receiver_entry);
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g_signal_connect (receiver_entry->webrtcbin, "on-ice-candidate",
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G_CALLBACK (on_ice_candidate_cb), (gpointer) receiver_entry);
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bus = gst_pipeline_get_bus (GST_PIPELINE (receiver_entry->pipeline));
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gst_bus_add_watch (bus, bus_watch_cb, receiver_entry->pipeline);
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gst_object_unref (bus);
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if (gst_element_set_state (receiver_entry->pipeline, GST_STATE_PLAYING) ==
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GST_STATE_CHANGE_FAILURE)
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g_error ("Could not start pipeline");
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return receiver_entry;
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cleanup:
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destroy_receiver_entry ((gpointer) receiver_entry);
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return NULL;
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}
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void
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destroy_receiver_entry (gpointer receiver_entry_ptr)
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{
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ReceiverEntry *receiver_entry = (ReceiverEntry *) receiver_entry_ptr;
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g_assert (receiver_entry != NULL);
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if (receiver_entry->pipeline != NULL) {
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GstBus *bus;
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gst_element_set_state (GST_ELEMENT (receiver_entry->pipeline),
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GST_STATE_NULL);
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bus = gst_pipeline_get_bus (GST_PIPELINE (receiver_entry->pipeline));
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gst_bus_remove_watch (bus);
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gst_object_unref (bus);
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gst_object_unref (GST_OBJECT (receiver_entry->webrtcbin));
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gst_object_unref (GST_OBJECT (receiver_entry->pipeline));
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}
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if (receiver_entry->connection != NULL)
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g_object_unref (G_OBJECT (receiver_entry->connection));
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g_slice_free1 (sizeof (ReceiverEntry), receiver_entry);
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}
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void
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on_offer_created_cb (GstPromise * promise, gpointer user_data)
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{
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gchar *sdp_string;
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gchar *json_string;
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JsonObject *sdp_json;
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JsonObject *sdp_data_json;
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GstStructure const *reply;
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GstPromise *local_desc_promise;
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GstWebRTCSessionDescription *offer = NULL;
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ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;
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reply = gst_promise_get_reply (promise);
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gst_structure_get (reply, "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
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&offer, NULL);
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gst_promise_unref (promise);
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local_desc_promise = gst_promise_new ();
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g_signal_emit_by_name (receiver_entry->webrtcbin, "set-local-description",
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offer, local_desc_promise);
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gst_promise_interrupt (local_desc_promise);
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gst_promise_unref (local_desc_promise);
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sdp_string = gst_sdp_message_as_text (offer->sdp);
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gst_print ("Negotiation offer created:\n%s\n", sdp_string);
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sdp_json = json_object_new ();
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json_object_set_string_member (sdp_json, "type", "sdp");
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sdp_data_json = json_object_new ();
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json_object_set_string_member (sdp_data_json, "type", "offer");
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json_object_set_string_member (sdp_data_json, "sdp", sdp_string);
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json_object_set_object_member (sdp_json, "data", sdp_data_json);
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json_string = get_string_from_json_object (sdp_json);
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json_object_unref (sdp_json);
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soup_websocket_connection_send_text (receiver_entry->connection, json_string);
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g_free (json_string);
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g_free (sdp_string);
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gst_webrtc_session_description_free (offer);
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}
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void
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on_negotiation_needed_cb (GstElement * webrtcbin, gpointer user_data)
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{
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GstPromise *promise;
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ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;
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gst_print ("Creating negotiation offer\n");
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promise = gst_promise_new_with_change_func (on_offer_created_cb,
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(gpointer) receiver_entry, NULL);
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g_signal_emit_by_name (G_OBJECT (webrtcbin), "create-offer", NULL, promise);
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}
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void
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on_ice_candidate_cb (G_GNUC_UNUSED GstElement * webrtcbin, guint mline_index,
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gchar * candidate, gpointer user_data)
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{
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JsonObject *ice_json;
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JsonObject *ice_data_json;
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gchar *json_string;
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ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;
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ice_json = json_object_new ();
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json_object_set_string_member (ice_json, "type", "ice");
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ice_data_json = json_object_new ();
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json_object_set_int_member (ice_data_json, "sdpMLineIndex", mline_index);
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json_object_set_string_member (ice_data_json, "candidate", candidate);
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json_object_set_object_member (ice_json, "data", ice_data_json);
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json_string = get_string_from_json_object (ice_json);
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json_object_unref (ice_json);
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soup_websocket_connection_send_text (receiver_entry->connection, json_string);
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g_free (json_string);
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}
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void
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soup_websocket_message_cb (G_GNUC_UNUSED SoupWebsocketConnection * connection,
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SoupWebsocketDataType data_type, GBytes * message, gpointer user_data)
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{
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gsize size;
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const gchar *data;
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gchar *data_string;
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const gchar *type_string;
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JsonNode *root_json;
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JsonObject *root_json_object;
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JsonObject *data_json_object;
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JsonParser *json_parser = NULL;
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ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;
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switch (data_type) {
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case SOUP_WEBSOCKET_DATA_BINARY:
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g_error ("Received unknown binary message, ignoring\n");
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return;
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case SOUP_WEBSOCKET_DATA_TEXT:
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data = g_bytes_get_data (message, &size);
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/* Convert to NULL-terminated string */
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data_string = g_strndup (data, size);
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break;
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default:
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g_assert_not_reached ();
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}
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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
|
|
|
|
|
|
static GOptionEntry entries[] = {
|
|
{"video-priority", 0, 0, G_OPTION_ARG_STRING, &video_priority,
|
|
"Priority of the video stream (very-low, low, medium or high)",
|
|
"PRIORITY"},
|
|
{"audio-priority", 0, 0, G_OPTION_ARG_STRING, &audio_priority,
|
|
"Priority of the audio stream (very-low, low, medium or high)",
|
|
"PRIORITY"},
|
|
{NULL},
|
|
};
|
|
|
|
int
|
|
main (int argc, char *argv[])
|
|
{
|
|
GMainLoop *mainloop;
|
|
SoupServer *soup_server;
|
|
GHashTable *receiver_entry_table;
|
|
GOptionContext *context;
|
|
GError *error = NULL;
|
|
|
|
setlocale (LC_ALL, "");
|
|
|
|
context = g_option_context_new ("- gstreamer webrtc sendonly 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)) {
|
|
g_printerr ("Error initializing: %s\n", error->message);
|
|
return -1;
|
|
}
|
|
|
|
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;
|
|
}
|