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

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Add a sendonly example
2020-01-14 23:47:27 +00:00
#include <locale.h>
#include <glib.h>
#include <gst/gst.h>
#include <gst/sdp/sdp.h>
sendonly: Don't assume we're building on UNIX Fixes https://github.com/centricular/gstwebrtc-demos/issues/203
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#ifdef G_OS_UNIX
#include <glib-unix.h>
#endif
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#define GST_USE_UNSTABLE_API
#include <gst/webrtc/webrtc.h>
#include <libsoup/soup.h>
#include <json-glib/json-glib.h>
#include <string.h>
#define RTP_PAYLOAD_TYPE "96"
webrtc sendonly: Add priority to example Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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#define RTP_AUDIO_PAYLOAD_TYPE "97"
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#define SOUP_HTTP_PORT 57778
#define STUN_SERVER "stun.l.google.com:19302"
webrtc: sendonly: Add support for Windows Add meson build script and use mfvideosrc element in case of Windows Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/29>
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#ifdef G_OS_WIN32
#define VIDEO_SRC "mfvideosrc"
#else
#define VIDEO_SRC "v4l2src"
#endif
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gchar *video_priority = NULL;
gchar *audio_priority = NULL;
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typedef struct _ReceiverEntry ReceiverEntry;
ReceiverEntry *create_receiver_entry (SoupWebsocketConnection * connection);
void destroy_receiver_entry (gpointer receiver_entry_ptr);
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 \
\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 \
webrtcPeerConnection.createAnswer().then(onLocalDescription).catch(reportError); \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 \
";
webrtc sendonly: Exit on bus errors Catch bus errors and cleanly error out Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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static gboolean
bus_watch_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
examples: webrtc: Add handling of the LATENCY messages to the C examples Without this the configured latency on the pipeline will be wrong. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609>
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GstPipeline *pipeline = user_data;
webrtc sendonly: Exit on bus errors Catch bus errors and cleanly error out Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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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;
}
examples: webrtc: Add handling of the LATENCY messages to the C examples Without this the configured latency on the pipeline will be wrong. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609>
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case GST_MESSAGE_LATENCY:
gst_bin_recalculate_latency (GST_BIN (pipeline));
break;
webrtc sendonly: Exit on bus errors Catch bus errors and cleanly error out Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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default:
break;
}
return G_SOURCE_CONTINUE;
}
webrtc sendonly: Add priority to example Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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static GstWebRTCPriorityType
_priority_from_string (const gchar * s)
{
GEnumClass *klass =
(GEnumClass *) g_type_class_ref (GST_TYPE_WEBRTC_PRIORITY_TYPE);
GEnumValue *en;
g_return_val_if_fail (klass, 0);
if (!(en = g_enum_get_value_by_name (klass, s)))
en = g_enum_get_value_by_nick (klass, s);
g_type_class_unref (klass);
if (en)
return en->value;
return 0;
}
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ReceiverEntry *
create_receiver_entry (SoupWebsocketConnection * connection)
{
GError *error;
ReceiverEntry *receiver_entry;
GstWebRTCRTPTransceiver *trans;
GArray *transceivers;
webrtc sendonly: Exit on bus errors Catch bus errors and cleanly error out Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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GstBus *bus;
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gst-examples: drop use of GSlice allocator Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
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receiver_entry = g_new0 (ReceiverEntry, 1);
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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;
gst-indent
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receiver_entry->pipeline =
gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://"
STUN_SERVER " "
webrtc: sendonly: Add support for Windows Add meson build script and use mfvideosrc element in case of Windows Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/29>
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VIDEO_SRC
" ! 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 ! "
webrtc: Change H264 examples to use aggregate-mode=zero-latency for best compatibility The default changed back to none because it broke existing code. See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/749 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/22>
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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="
webrtc sendonly: Add priority to example Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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RTP_PAYLOAD_TYPE " ! webrtcbin. "
webrtc examples: Force regular non-MULTIOPUS Using MULTIOPUS breaks with most browsers Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3675>
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"autoaudiosrc ! queue max-size-buffers=1 leaky=downstream"
examples: set perfect-timestamp=true on opusenc Fix audio streaming on Chrome, see https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6512>
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" ! audioconvert ! audioresample ! opusenc perfect-timestamp=true ! rtpopuspay pt="
webrtc examples: Force regular non-MULTIOPUS Using MULTIOPUS breaks with most browsers Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3675>
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RTP_AUDIO_PAYLOAD_TYPE " ! application/x-rtp, encoding-name=OPUS !"
" webrtcbin. ", &error);
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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);
gst-indent
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g_signal_emit_by_name (receiver_entry->webrtcbin, "get-transceivers",
&transceivers);
webrtc sendonly: Add priority to example Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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g_assert (transceivers != NULL && transceivers->len > 1);
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trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);
webrtc: Use properties to access the inside of the transceiver object This will allow hiding the insides from unsafe application access. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/36>
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g_object_set (trans, "direction",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, NULL);
webrtc sendonly: Add priority to example Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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if (video_priority) {
GstWebRTCPriorityType priority;
priority = _priority_from_string (video_priority);
webrtc: Use properties to access the inside of the transceiver object This will allow hiding the insides from unsafe application access. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/36>
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if (priority) {
GstWebRTCRTPSender *sender;
g_object_get (trans, "sender", &sender, NULL);
gst_webrtc_rtp_sender_set_priority (sender, priority);
g_object_unref (sender);
}
webrtc sendonly: Add priority to example Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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}
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
webrtc: Use properties to access the inside of the transceiver object This will allow hiding the insides from unsafe application access. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/36>
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g_object_set (trans, "direction",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, NULL);
webrtc sendonly: Add priority to example Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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if (audio_priority) {
GstWebRTCPriorityType priority;
priority = _priority_from_string (audio_priority);
webrtc: Use properties to access the inside of the transceiver object This will allow hiding the insides from unsafe application access. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/36>
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if (priority) {
GstWebRTCRTPSender *sender;
g_object_get (trans, "sender", &sender, NULL);
gst_webrtc_rtp_sender_set_priority (sender, priority);
g_object_unref (sender);
}
webrtc sendonly: Add priority to example Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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}
sendonly: Fix transceivers leak. Make sure to unref the transceivers array after use.
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g_array_unref (transceivers);
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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);
webrtc sendonly: Exit on bus errors Catch bus errors and cleanly error out Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
2020-07-09 20:30:41 +00:00
bus = gst_pipeline_get_bus (GST_PIPELINE (receiver_entry->pipeline));
examples: webrtc: Add handling of the LATENCY messages to the C examples Without this the configured latency on the pipeline will be wrong. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609>
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gst_bus_add_watch (bus, bus_watch_cb, receiver_entry->pipeline);
webrtc sendonly: Exit on bus errors Catch bus errors and cleanly error out Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
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gst_object_unref (bus);
if (gst_element_set_state (receiver_entry->pipeline, GST_STATE_PLAYING) ==
GST_STATE_CHANGE_FAILURE)
g_error ("Could not start pipeline");
Add a sendonly example
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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) {
examples: webrtc: Remove the bus watch at the end Otherwise a file descriptor will be leaked. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609>
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GstBus *bus;
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gst_element_set_state (GST_ELEMENT (receiver_entry->pipeline),
GST_STATE_NULL);
examples: webrtc: Remove the bus watch at the end Otherwise a file descriptor will be leaked. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609>
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bus = gst_pipeline_get_bus (GST_PIPELINE (receiver_entry->pipeline));
gst_bus_remove_watch (bus);
gst_object_unref (bus);
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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));
gst-examples: drop use of GSlice allocator Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
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g_free (receiver_entry);
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}
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);
Port to gst_print* family g_print* would print broken string on Windows See also https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/258 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/20>
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gst_print ("Negotiation offer created:\n%s\n", sdp_string);
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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;
Port to gst_print* family g_print* would print broken string on Windows See also https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/258 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/20>
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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,
(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;
webrtc-unidirectional: Avoid critical Don't unref the parameter passed to a signal, it's always owned by the caller. Fixes a GLib critical. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3675>
2023-01-03 22:12:12 +00:00
const gchar *data;
Add a sendonly example
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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:
webrtc-unidirectional: Avoid critical Don't unref the parameter passed to a signal, it's always owned by the caller. Fixes a GLib critical. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3675>
2023-01-03 22:12:12 +00:00
data = g_bytes_get_data (message, &size);
Add a sendonly example
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/* 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");
Port to gst_print* family g_print* would print broken string on Windows See also https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/258 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/20>
2020-07-26 17:20:59 +00:00
gst_print ("Received SDP:\n%s\n", sdp_string);
Add a sendonly example
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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");
Port to gst_print* family g_print* would print broken string on Windows See also https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/258 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/20>
2020-07-26 17:20:59 +00:00
gst_print ("Received ICE candidate with mline index %u; candidate: %s\n",
Add a sendonly example
2020-01-14 23:47:27 +00:00
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);
Port to gst_print* family g_print* would print broken string on Windows See also https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/258 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/20>
2020-07-26 17:20:59 +00:00
gst_print ("Closed websocket connection %p\n", (gpointer) connection);
Add a sendonly example
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}
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;
Port to gst_print* family g_print* would print broken string on Windows See also https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/258 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/20>
2020-07-26 17:20:59 +00:00
gst_print ("Processing new websocket connection %p", (gpointer) connection);
Add a sendonly example
2020-01-14 23:47:27 +00:00
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;
}
sendonly: Don't assume we're building on UNIX Fixes https://github.com/centricular/gstwebrtc-demos/issues/203
2020-06-22 12:09:12 +00:00
#ifdef G_OS_UNIX
Add a sendonly example
2020-01-14 23:47:27 +00:00
gboolean
exit_sighandler (gpointer user_data)
{
Port to gst_print* family g_print* would print broken string on Windows See also https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/258 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/20>
2020-07-26 17:20:59 +00:00
gst_print ("Caught signal, stopping mainloop\n");
Add a sendonly example
2020-01-14 23:47:27 +00:00
GMainLoop *mainloop = (GMainLoop *) user_data;
g_main_loop_quit (mainloop);
return TRUE;
}
sendonly: Don't assume we're building on UNIX Fixes https://github.com/centricular/gstwebrtc-demos/issues/203
2020-06-22 12:09:12 +00:00
#endif
Add a sendonly example
2020-01-14 23:47:27 +00:00
webrtc sendonly: Add priority to example Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
2020-07-09 21:07:10 +00:00
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},
};
Add a sendonly example
2020-01-14 23:47:27 +00:00
int
main (int argc, char *argv[])
{
GMainLoop *mainloop;
SoupServer *soup_server;
GHashTable *receiver_entry_table;
webrtc sendonly: Add priority to example Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
2020-07-09 21:07:10 +00:00
GOptionContext *context;
GError *error = NULL;
Add a sendonly example
2020-01-14 23:47:27 +00:00
setlocale (LC_ALL, "");
webrtc sendonly: Add priority to example Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
2020-07-09 21:07:10 +00:00
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;
}
Add a sendonly example
2020-01-14 23:47:27 +00:00
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);
sendonly: Don't assume we're building on UNIX Fixes https://github.com/centricular/gstwebrtc-demos/issues/203
2020-06-22 12:09:12 +00:00
#ifdef G_OS_UNIX
Add a sendonly example
2020-01-14 23:47:27 +00:00
g_unix_signal_add (SIGINT, exit_sighandler, mainloop);
g_unix_signal_add (SIGTERM, exit_sighandler, mainloop);
sendonly: Don't assume we're building on UNIX Fixes https://github.com/centricular/gstwebrtc-demos/issues/203
2020-06-22 12:09:12 +00:00
#endif
Add a sendonly example
2020-01-14 23:47:27 +00:00
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);
Port to gst_print* family g_print* would print broken string on Windows See also https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/258 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/20>
2020-07-26 17:20:59 +00:00
gst_print ("WebRTC page link: http://127.0.0.1:%d/\n", (gint) SOUP_HTTP_PORT);
Add a sendonly example
2020-01-14 23:47:27 +00:00
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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