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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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58
webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
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@ -17,23 +17,24 @@
#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 */
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;