From 804c0c2f5ead8264414e2a9b0cb46dc42124e56d Mon Sep 17 00:00:00 2001 From: Costa Shulyupin Date: Tue, 14 Apr 2020 13:49:41 +0300 Subject: [PATCH] gst-indent --- .../gst/mp-webrtc-sendrecv.c | 58 ++++++++++--------- 1 file changed, 30 insertions(+), 28 deletions(-) diff --git a/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c b/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c index d1d3561bdb..6e66fbd162 100644 --- a/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c +++ b/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c @@ -17,23 +17,24 @@ #include -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 */ ROOM_JOINING = 3000, ROOM_JOIN_ERROR, ROOM_JOINED, ROOM_CALL_NEGOTIATING = 4000, /* negotiating with some or all peers */ - ROOM_CALL_OFFERING, /* when we're the one sending the offer */ - ROOM_CALL_ANSWERING, /* when we're the one answering an offer */ - ROOM_CALL_STARTED, /* in a call with some or all peers */ + ROOM_CALL_OFFERING, /* when we're the one sending the offer */ + ROOM_CALL_ANSWERING, /* when we're the one answering an offer */ + ROOM_CALL_STARTED, /* in a call with some or all peers */ ROOM_CALL_STOPPING, ROOM_CALL_STOPPED, ROOM_CALL_ERROR, @@ -51,12 +52,14 @@ static gchar *local_id = NULL; static gchar *room_id = NULL; static gboolean strict_ssl = TRUE; -static GOptionEntry entries[] = -{ - { "name", 0, 0, G_OPTION_ARG_STRING, &local_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" }, - { "server", 0, 0, G_OPTION_ARG_STRING, &server_url, "Signalling server to connect to", "URL" }, - { NULL } +static GOptionEntry entries[] = { + {"name", 0, 0, G_OPTION_ARG_STRING, &local_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"}, + {"server", 0, 0, G_OPTION_ARG_STRING, &server_url, + "Signalling server to connect to", "URL"}, + {NULL} }; static gint @@ -97,7 +100,7 @@ cleanup_and_quit_loop (const gchar * msg, enum AppState state) return G_SOURCE_REMOVE; } -static gchar* +static gchar * get_string_from_json_object (JsonObject * object) { JsonNode *root; @@ -117,8 +120,8 @@ get_string_from_json_object (JsonObject * object) } 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, *sink; @@ -415,8 +418,7 @@ start_pipeline (void) * we can preroll early. */ pipeline = gst_parse_launch ("tee name=audiotee ! queue ! fakesink " "audiotestsrc is-live=true wave=red-noise ! queue ! opusenc ! rtpopuspay ! " - "queue ! " RTP_CAPS_OPUS(96) " ! audiotee. ", - &error); + "queue ! " RTP_CAPS_OPUS (96) " ! audiotee. ", &error); if (error) { 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"); /* Start recording, but not transmitting */ if (!start_pipeline ()) { - cleanup_and_quit_loop ("ERROR: Failed to start pipeline", - ROOM_CALL_ERROR); + cleanup_and_quit_loop ("ERROR: Failed to start pipeline", ROOM_CALL_ERROR); return; } @@ -767,7 +768,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type, 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 */ @@ -782,7 +783,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type, if (g_strcmp0 (text, "HELLO") == 0) { /* May fail asynchronously */ do_registration (); - /* Room-related message */ + /* Room-related message */ } else if (g_str_has_prefix (text, "ROOM_")) { /* Room joined, now we can start negotiation */ if (g_str_has_prefix (text, "ROOM_OK ")) { @@ -811,7 +812,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type, peers = g_list_remove (peers, peer_id); g_print ("Peer %s has left the room\n", peer_id); remove_peer_from_pipeline (peer_id); - g_free ((gchar*) peer_id); + g_free ((gchar *) peer_id); /* TODO: cleanup pipeline */ } else { g_printerr ("WARNING: Ignoring unknown message %s\n", text); @@ -820,7 +821,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type, } else { goto err; } - /* Handle errors */ + /* Handle errors */ } else if (g_str_has_prefix (text, "ERROR")) { handle_error_message (text); } else { @@ -842,7 +843,7 @@ err: static void on_server_connected (SoupSession * session, GAsyncResult * res, - SoupMessage *msg) + SoupMessage * msg) { GError *error = NULL; @@ -875,7 +876,7 @@ connect_to_websocket_server_async (void) 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, strict_ssl, SOUP_SESSION_SSL_USE_SYSTEM_CA_FILE, TRUE, @@ -904,7 +905,8 @@ check_plugins (void) GstPlugin *plugin; GstRegistry *registry; const gchar *needed[] = { "opus", "nice", "webrtc", "dtls", "srtp", - "rtpmanager", "audiotestsrc", NULL}; + "rtpmanager", "audiotestsrc", NULL + }; registry = gst_registry_get (); ret = TRUE;