#include <string.h>
-enum AppState {
+enum AppState
+{
APP_STATE_UNKNOWN = 0,
- APP_STATE_ERROR = 1, /* generic error */
+ APP_STATE_ERROR = 1, /* generic error */
SERVER_CONNECTING = 1000,
SERVER_CONNECTION_ERROR,
- SERVER_CONNECTED, /* Ready to register */
+ SERVER_CONNECTED, /* Ready to register */
SERVER_REGISTERING = 2000,
SERVER_REGISTRATION_ERROR,
- SERVER_REGISTERED, /* Ready to call a peer */
- SERVER_CLOSED, /* server connection closed by us or the server */
+ SERVER_REGISTERED, /* Ready to call a peer */
+ SERVER_CLOSED, /* server connection closed by us or the server */
PEER_CONNECTING = 3000,
PEER_CONNECTION_ERROR,
PEER_CONNECTED,
static gboolean disable_ssl = FALSE;
static gboolean remote_is_offerer = FALSE;
-static GOptionEntry entries[] =
-{
- { "peer-id", 0, 0, G_OPTION_ARG_STRING, &peer_id, "String ID of the peer to connect to", "ID" },
- { "server", 0, 0, G_OPTION_ARG_STRING, &server_url, "Signalling server to connect to", "URL" },
- { "disable-ssl", 0, 0, G_OPTION_ARG_NONE, &disable_ssl, "Disable ssl", NULL },
- { "remote-offerer", 0, 0, G_OPTION_ARG_NONE, &remote_is_offerer, "Request that the peer generate the offer and we'll answer", NULL },
- { NULL },
+static GOptionEntry entries[] = {
+ {"peer-id", 0, 0, G_OPTION_ARG_STRING, &peer_id,
+ "String ID of the peer to connect to", "ID"},
+ {"server", 0, 0, G_OPTION_ARG_STRING, &server_url,
+ "Signalling server to connect to", "URL"},
+ {"disable-ssl", 0, 0, G_OPTION_ARG_NONE, &disable_ssl, "Disable ssl", NULL},
+ {"remote-offerer", 0, 0, G_OPTION_ARG_NONE, &remote_is_offerer,
+ "Request that the peer generate the offer and we'll answer", NULL},
+ {NULL},
};
static gboolean
return G_SOURCE_REMOVE;
}
-static gchar*
+static gchar *
get_string_from_json_object (JsonObject * object)
{
JsonNode *root;
}
static void
-handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name,
- const char * sink_name)
+handle_media_stream (GstPad * pad, GstElement * pipe, const char *convert_name,
+ const char *sink_name)
{
GstPad *qpad;
GstElement *q, *conv, *resample, *sink;
}
static void
-send_sdp_to_peer (GstWebRTCSessionDescription *desc)
+send_sdp_to_peer (GstWebRTCSessionDescription * desc)
{
gchar *text;
JsonObject *msg, *sdp;
if (app_state < PEER_CALL_NEGOTIATING) {
- cleanup_and_quit_loop ("Can't send SDP to peer, not in call", APP_STATE_ERROR);
+ cleanup_and_quit_loop ("Can't send SDP to peer, not in call",
+ APP_STATE_ERROR);
return;
}
if (desc->type == GST_WEBRTC_SDP_TYPE_OFFER) {
g_print ("Sending offer:\n%s\n", text);
json_object_set_string_member (sdp, "type", "offer");
- }
- else if (desc->type == GST_WEBRTC_SDP_TYPE_ANSWER) {
+ } else if (desc->type == GST_WEBRTC_SDP_TYPE_ANSWER) {
g_print ("Sending answer:\n%s\n", text);
json_object_set_string_member (sdp, "type", "answer");
- }
- else {
+ } else {
g_assert_not_reached ();
}
g_assert_cmphex (app_state, ==, PEER_CALL_NEGOTIATING);
- g_assert_cmphex (gst_promise_wait(promise), ==, GST_PROMISE_RESULT_REPLIED);
+ g_assert_cmphex (gst_promise_wait (promise), ==, GST_PROMISE_RESULT_REPLIED);
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "offer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
g_free (msg);
} else {
GstPromise *promise;
- promise = gst_promise_new_with_change_func (on_offer_created, user_data, NULL);;
+ promise =
+ gst_promise_new_with_change_func (on_offer_created, user_data, NULL);;
g_signal_emit_by_name (webrtc1, "create-offer", NULL, promise);
}
}
static void
data_channel_on_open (GObject * dc, gpointer user_data)
{
- GBytes *bytes = g_bytes_new ("data", strlen("data"));
+ GBytes *bytes = g_bytes_new ("data", strlen ("data"));
g_print ("data channel opened\n");
g_signal_emit_by_name (dc, "send-string", "Hi! from GStreamer");
g_signal_emit_by_name (dc, "send-data", bytes);
}
static void
-data_channel_on_message_string (GObject * dc, gchar *str, gpointer user_data)
+data_channel_on_message_string (GObject * dc, gchar * str, gpointer user_data)
{
g_print ("Received data channel message: %s\n", str);
}
static void
connect_data_channel_signals (GObject * data_channel)
{
- g_signal_connect (data_channel, "on-error", G_CALLBACK (data_channel_on_error),
- NULL);
+ g_signal_connect (data_channel, "on-error",
+ G_CALLBACK (data_channel_on_error), NULL);
g_signal_connect (data_channel, "on-open", G_CALLBACK (data_channel_on_open),
NULL);
- g_signal_connect (data_channel, "on-close", G_CALLBACK (data_channel_on_close),
- NULL);
- g_signal_connect (data_channel, "on-message-string", G_CALLBACK (data_channel_on_message_string),
- NULL);
+ g_signal_connect (data_channel, "on-close",
+ G_CALLBACK (data_channel_on_close), NULL);
+ g_signal_connect (data_channel, "on-message-string",
+ G_CALLBACK (data_channel_on_message_string), NULL);
}
static void
-on_data_channel (GstElement * webrtc, GObject * data_channel, gpointer user_data)
+on_data_channel (GstElement * webrtc, GObject * data_channel,
+ gpointer user_data)
{
connect_data_channel_signals (data_channel);
receive_channel = data_channel;
GstWebRTCICEGatheringState ice_gather_state;
const gchar *new_state = "unknown";
- g_object_get (webrtcbin, "ice-gathering-state", &ice_gather_state,
- NULL);
+ g_object_get (webrtcbin, "ice-gathering-state", &ice_gather_state, NULL);
switch (ice_gather_state) {
case GST_WEBRTC_ICE_GATHERING_STATE_NEW:
new_state = "new";
GError *error = NULL;
pipe1 =
- gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv " STUN_SERVER
+ gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv "
+ STUN_SERVER
"videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! "
"queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
"audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
- "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ",
- &error);
+ "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ", &error);
if (error) {
g_printerr ("Failed to parse launch: %s\n", error->message);
g_assert_cmphex (app_state, ==, PEER_CALL_NEGOTIATING);
- g_assert_cmphex (gst_promise_wait(promise), ==, GST_PROMISE_RESULT_REPLIED);
+ g_assert_cmphex (gst_promise_wait (promise), ==, GST_PROMISE_RESULT_REPLIED);
reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "answer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &answer, NULL);
}
static void
-on_offer_received (GstSDPMessage *sdp)
+on_offer_received (GstSDPMessage * sdp)
{
GstWebRTCSessionDescription *offer = NULL;
GstPromise *promise;
/* Set remote description on our pipeline */
{
promise = gst_promise_new ();
- g_signal_emit_by_name (webrtc1, "set-remote-description", offer,
- promise);
+ g_signal_emit_by_name (webrtc1, "set-remote-description", offer, promise);
gst_promise_interrupt (promise);
gst_promise_unref (promise);
}
gst_webrtc_session_description_free (offer);
- promise = gst_promise_new_with_change_func (on_answer_created, NULL,
- NULL);
+ promise = gst_promise_new_with_change_func (on_answer_created, NULL, NULL);
g_signal_emit_by_name (webrtc1, "create-answer", NULL, promise);
}
case SOUP_WEBSOCKET_DATA_BINARY:
g_printerr ("Received unknown binary message, ignoring\n");
return;
- case SOUP_WEBSOCKET_DATA_TEXT: {
+ case SOUP_WEBSOCKET_DATA_TEXT:{
gsize size;
const gchar *data = g_bytes_get_data (message, &size);
/* Convert to NULL-terminated string */
cleanup_and_quit_loop ("ERROR: Failed to setup call", PEER_CALL_ERROR);
goto out;
}
- /* Call has been setup by the server, now we can start negotiation */
+ /* Call has been setup by the server, now we can start negotiation */
} else if (g_strcmp0 (text, "SESSION_OK") == 0) {
if (app_state != PEER_CONNECTING) {
cleanup_and_quit_loop ("ERROR: Received SESSION_OK when not calling",
if (!start_pipeline ())
cleanup_and_quit_loop ("ERROR: failed to start pipeline",
PEER_CALL_ERROR);
- /* Handle errors */
+ /* Handle errors */
} else if (g_str_has_prefix (text, "ERROR")) {
switch (app_state) {
case SERVER_CONNECTING:
app_state = APP_STATE_ERROR;
}
cleanup_and_quit_loop (text, 0);
- /* Look for JSON messages containing SDP and ICE candidates */
+ /* Look for JSON messages containing SDP and ICE candidates */
} else {
JsonNode *root;
JsonObject *object, *child;
answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER,
sdp);
g_assert_nonnull (answer);
-
+
/* Set remote description on our pipeline */
{
GstPromise *promise = gst_promise_new ();
gst_promise_unref (promise);
}
app_state = PEER_CALL_STARTED;
- }
- else {
+ } else {
g_print ("Received offer:\n%s\n", text);
on_offer_received (sdp);
}
static void
on_server_connected (SoupSession * session, GAsyncResult * res,
- SoupMessage *msg)
+ SoupMessage * msg)
{
GError *error = NULL;
SoupLogger *logger;
SoupMessage *message;
SoupSession *session;
- const char *https_aliases[] = {"wss", NULL};
+ const char *https_aliases[] = { "wss", NULL };
- session = soup_session_new_with_options (SOUP_SESSION_SSL_STRICT, !disable_ssl,
+ session =
+ soup_session_new_with_options (SOUP_SESSION_SSL_STRICT, !disable_ssl,
SOUP_SESSION_SSL_USE_SYSTEM_CA_FILE, TRUE,
//SOUP_SESSION_SSL_CA_FILE, "/etc/ssl/certs/ca-bundle.crt",
SOUP_SESSION_HTTPS_ALIASES, https_aliases, NULL);
GstPlugin *plugin;
GstRegistry *registry;
const gchar *needed[] = { "opus", "vpx", "nice", "webrtc", "dtls", "srtp",
- "rtpmanager", "videotestsrc", "audiotestsrc", NULL};
+ "rtpmanager", "videotestsrc", "audiotestsrc", NULL
+ };
registry = gst_registry_get ();
ret = TRUE;
projects / platform / upstream / gstreamer.git / commitdiff