mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-13 11:56:35 +00:00
255fef3896
This example sets up a recvonly H.264 transceiver and receives H.264 from a peer, while sending bi-directional Opus audio.
702 lines
22 KiB
C
702 lines
22 KiB
C
#include <locale.h>
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#include <glib.h>
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#include <glib-unix.h>
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#include <gst/gst.h>
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#include <gst/sdp/sdp.h>
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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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/* This example is a standalone app which serves a web page
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* and configures webrtcbin to receive an H.264 video feed, and to
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* send+recv an Opus audio stream */
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#define RTP_PAYLOAD_TYPE "96"
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#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
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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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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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GstPadProbeReturn payloader_caps_event_probe_cb (GstPad * pad,
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GstPadProbeInfo * info, gpointer user_data);
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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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gboolean exit_sighandler (gpointer user_data);
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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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function getLocalStream() { \n \
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var constraints = {\"video\":true,\"audio\":true}; \n \
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if (navigator.mediaDevices.getUserMedia) { \n \
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return navigator.mediaDevices.getUserMedia(constraints); \n \
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} \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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/* Send our video/audio to the other peer */ \n \
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local_stream_promise = getLocalStream().then((stream) => { \n \
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console.log('Adding local stream'); \n \
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webrtcPeerConnection.addStream(stream); \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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\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 void
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handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name,
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const char * sink_name)
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{
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GstPad *qpad;
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GstElement *q, *conv, *resample, *sink;
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GstPadLinkReturn ret;
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g_print ("Trying to handle stream with %s ! %s", convert_name, sink_name);
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q = gst_element_factory_make ("queue", NULL);
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g_assert_nonnull (q);
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conv = gst_element_factory_make (convert_name, NULL);
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g_assert_nonnull (conv);
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sink = gst_element_factory_make (sink_name, NULL);
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g_assert_nonnull (sink);
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if (g_strcmp0 (convert_name, "audioconvert") == 0) {
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/* Might also need to resample, so add it just in case.
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* Will be a no-op if it's not required. */
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resample = gst_element_factory_make ("audioresample", NULL);
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g_assert_nonnull (resample);
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gst_bin_add_many (GST_BIN (pipe), q, conv, resample, sink, NULL);
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gst_element_sync_state_with_parent (q);
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gst_element_sync_state_with_parent (conv);
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gst_element_sync_state_with_parent (resample);
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gst_element_sync_state_with_parent (sink);
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gst_element_link_many (q, conv, resample, sink, NULL);
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} else {
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gst_bin_add_many (GST_BIN (pipe), q, conv, sink, NULL);
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gst_element_sync_state_with_parent (q);
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gst_element_sync_state_with_parent (conv);
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gst_element_sync_state_with_parent (sink);
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gst_element_link_many (q, conv, sink, NULL);
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}
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qpad = gst_element_get_static_pad (q, "sink");
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ret = gst_pad_link (pad, qpad);
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g_assert_cmphex (ret, ==, GST_PAD_LINK_OK);
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}
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static void
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on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
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GstElement * pipe)
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{
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GstCaps *caps;
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const gchar *name;
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if (!gst_pad_has_current_caps (pad)) {
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g_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
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GST_PAD_NAME (pad));
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return;
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}
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caps = gst_pad_get_current_caps (pad);
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name = gst_structure_get_name (gst_caps_get_structure (caps, 0));
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if (g_str_has_prefix (name, "video")) {
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handle_media_stream (pad, pipe, "videoconvert", "autovideosink");
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} else if (g_str_has_prefix (name, "audio")) {
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handle_media_stream (pad, pipe, "audioconvert", "autoaudiosink");
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} else {
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g_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad));
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}
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}
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static void
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on_incoming_stream (GstElement * webrtc, GstPad * pad, ReceiverEntry *receiver_entry)
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{
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GstElement *decodebin;
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GstPad *sinkpad;
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if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC)
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return;
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decodebin = gst_element_factory_make ("decodebin", NULL);
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g_signal_connect (decodebin, "pad-added",
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G_CALLBACK (on_incoming_decodebin_stream), receiver_entry->pipeline);
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gst_bin_add (GST_BIN (receiver_entry->pipeline), decodebin);
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gst_element_sync_state_with_parent (decodebin);
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sinkpad = gst_element_get_static_pad (decodebin, "sink");
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gst_pad_link (pad, sinkpad);
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gst_object_unref (sinkpad);
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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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GstCaps *video_caps;
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GstWebRTCRTPTransceiver *trans = NULL;
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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 = gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://" STUN_SERVER " "
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"audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
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"queue ! " RTP_CAPS_OPUS "97 ! webrtcbin. "
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, &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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/* Incoming streams will be exposed via this signal */
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g_signal_connect (receiver_entry->webrtcbin, "pad-added", G_CALLBACK (on_incoming_stream),
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receiver_entry);
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#if 0
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GstElement *rtpbin = gst_bin_get_by_name (GST_BIN (receiver_entry->webrtcbin), "rtpbin");
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g_object_set (rtpbin, "latency", 40, NULL);
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gst_object_unref (rtpbin);
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#endif
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// Create a 2nd transceiver for the receive only video stream
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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");
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g_signal_emit_by_name (receiver_entry->webrtcbin, "add-transceiver", GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, video_caps, NULL, &trans);
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gst_caps_unref (video_caps);
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gst_object_unref (trans);
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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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gst_element_set_state (receiver_entry->pipeline, GST_STATE_PLAYING);
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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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gst_element_set_state (GST_ELEMENT (receiver_entry->pipeline),
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GST_STATE_NULL);
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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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g_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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g_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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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");
|
|
g_bytes_unref (message);
|
|
return;
|
|
|
|
case SOUP_WEBSOCKET_DATA_TEXT:
|
|
data = g_bytes_unref_to_data (message, &size);
|
|
/* Convert to NULL-terminated string */
|
|
data_string = g_strndup (data, size);
|
|
g_free (data);
|
|
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");
|
|
|
|
g_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");
|
|
|
|
g_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);
|
|
g_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;
|
|
|
|
g_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;
|
|
}
|
|
|
|
|
|
gboolean
|
|
exit_sighandler (gpointer user_data)
|
|
{
|
|
g_print ("Caught signal, stopping mainloop\n");
|
|
GMainLoop *mainloop = (GMainLoop *) user_data;
|
|
g_main_loop_quit (mainloop);
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
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);
|
|
|
|
g_unix_signal_add (SIGINT, exit_sighandler, mainloop);
|
|
g_unix_signal_add (SIGTERM, exit_sighandler, mainloop);
|
|
|
|
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);
|
|
|
|
g_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;
|
|
}
|