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gst-indent

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Costa Shulyupin 2020-04-14 13:49:55 +03:00 committed by Sebastian Dröge
parent ca96b6de86
commit 2557eab9d5
1 changed files with 56 additions and 54 deletions
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110
webrtc/sendrecv/gst/webrtc-sendrecv.c
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@ -18,16 +18,17 @@
#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,
@ -49,13 +50,15 @@ static const gchar *server_url = "wss://webrtc.nirbheek.in:8443";
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
@ -84,7 +87,7 @@ 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;
@ -104,8 +107,8 @@ 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, *resample, *sink;
@ -215,13 +218,14 @@ send_ice_candidate_message (GstElement * webrtc G_GNUC_UNUSED, guint mlineindex,
}
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;
}
@ -231,12 +235,10 @@ send_sdp_to_peer (GstWebRTCSessionDescription *desc)
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 ();
}
@ -261,7 +263,7 @@ on_offer_created (GstPromise * promise, gpointer user_data)
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);
@ -288,7 +290,8 @@ on_negotiation_needed (GstElement * element, gpointer user_data)
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);
}
}
@ -306,7 +309,7 @@ data_channel_on_error (GObject * dc, gpointer user_data)
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);
@ -320,7 +323,7 @@ data_channel_on_close (GObject * dc, gpointer user_data)
}
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);
}
@ -328,18 +331,19 @@ data_channel_on_message_string (GObject * dc, gchar *str, gpointer user_data)
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;
@ -352,8 +356,7 @@ on_ice_gathering_state_notify (GstElement * webrtcbin, GParamSpec * pspec,
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";
@ -375,12 +378,12 @@ start_pipeline (void)
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);
@ -497,7 +500,7 @@ on_answer_created (GstPromise * promise, gpointer user_data)
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);
@ -514,7 +517,7 @@ on_answer_created (GstPromise * promise, gpointer user_data)
}
static void
on_offer_received (GstSDPMessage *sdp)
on_offer_received (GstSDPMessage * sdp)
{
GstWebRTCSessionDescription *offer = NULL;
GstPromise *promise;
@ -525,15 +528,13 @@ on_offer_received (GstSDPMessage *sdp)
/* 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);
}
@ -548,7 +549,7 @@ 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 */
@ -573,7 +574,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
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",
@ -586,7 +587,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
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:
@ -605,7 +606,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
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;
@ -659,7 +660,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
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 ();
@ -669,8 +670,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
gst_promise_unref (promise);
}
app_state = PEER_CALL_STARTED;
}
else {
} else {
g_print ("Received offer:\n%s\n", text);
on_offer_received (sdp);
}
@ -698,7 +698,7 @@ out:
static void
on_server_connected (SoupSession * session, GAsyncResult * res,
SoupMessage *msg)
SoupMessage * msg)
{
GError *error = NULL;
@ -730,9 +730,10 @@ 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, !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);
@ -759,7 +760,8 @@ check_plugins (void)
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;