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gst-indent

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Costa Shulyupin 2020-04-14 13:49:41 +03:00 committed by Sebastian Dröge
parent 65db695212
commit 804c0c2f5e
1 changed files with 30 additions and 28 deletions
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24
webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
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@ -17,7 +17,8 @@
#include <string.h> #include <string.h>
enum AppState { enum AppState
{
APP_STATE_UNKNOWN = 0, APP_STATE_UNKNOWN = 0,
APP_STATE_ERROR = 1, /* generic error */ APP_STATE_ERROR = 1, /* generic error */
SERVER_CONNECTING = 1000, SERVER_CONNECTING = 1000,
@ -51,11 +52,13 @@ static gchar *local_id = NULL;
static gchar *room_id = NULL; static gchar *room_id = NULL;
static gboolean strict_ssl = TRUE; static gboolean strict_ssl = TRUE;
static GOptionEntry entries[] = static GOptionEntry entries[] = {
{ {"name", 0, 0, G_OPTION_ARG_STRING, &local_id,
{ "name", 0, 0, G_OPTION_ARG_STRING, &local_id, "Name we will send to the server", "ID" }, "Name we will send to the server", "ID"},
{ "room-id", 0, 0, G_OPTION_ARG_STRING, &room_id, "Room name to join or create", "ID" }, {"room-id", 0, 0, G_OPTION_ARG_STRING, &room_id,
{ "server", 0, 0, G_OPTION_ARG_STRING, &server_url, "Signalling server to connect to", "URL" }, "Room name to join or create", "ID"},
{"server", 0, 0, G_OPTION_ARG_STRING, &server_url,
"Signalling server to connect to", "URL"},
{NULL} {NULL}
}; };
@ -415,8 +418,7 @@ start_pipeline (void)
* we can preroll early. */ * we can preroll early. */
pipeline = gst_parse_launch ("tee name=audiotee ! queue ! fakesink " pipeline = gst_parse_launch ("tee name=audiotee ! queue ! fakesink "
"audiotestsrc is-live=true wave=red-noise ! queue ! opusenc ! rtpopuspay ! " "audiotestsrc is-live=true wave=red-noise ! queue ! opusenc ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS(96) " ! audiotee. ", "queue ! " RTP_CAPS_OPUS (96) " ! audiotee. ", &error);
&error);
if (error) { if (error) {
g_printerr ("Failed to parse launch: %s\n", error->message); g_printerr ("Failed to parse launch: %s\n", error->message);
@ -525,8 +527,7 @@ do_join_room (const gchar * text)
g_print ("Room joined\n"); g_print ("Room joined\n");
/* Start recording, but not transmitting */ /* Start recording, but not transmitting */
if (!start_pipeline ()) { if (!start_pipeline ()) {
cleanup_and_quit_loop ("ERROR: Failed to start pipeline", cleanup_and_quit_loop ("ERROR: Failed to start pipeline", ROOM_CALL_ERROR);
ROOM_CALL_ERROR);
return; return;
} }
@ -904,7 +905,8 @@ check_plugins (void)
GstPlugin *plugin; GstPlugin *plugin;
GstRegistry *registry; GstRegistry *registry;
const gchar *needed[] = { "opus", "nice", "webrtc", "dtls", "srtp", const gchar *needed[] = { "opus", "nice", "webrtc", "dtls", "srtp",
"rtpmanager", "audiotestsrc", NULL}; "rtpmanager", "audiotestsrc", NULL
};
registry = gst_registry_get (); registry = gst_registry_get ();
ret = TRUE; ret = TRUE;