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Fix crash on Windows by delimiting option entries with NULL

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Also use more verbose forms of g_assert which print values on failure
This commit is contained in:
Nirbheek Chauhan 2018-03-08 20:10:55 +05:30
parent 492d13a7c9
commit fa2adc717b
2 changed files with 17 additions and 13 deletions
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1
webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
View file

@ -54,6 +54,7 @@ static GOptionEntry entries[] =
{ "name", 0, 0, G_OPTION_ARG_STRING, &local_id, "Name we will send to the server", "ID" }, { "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" }, { "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" }, { "server", 0, 0, G_OPTION_ARG_STRING, &server_url, "Signalling server to connect to", "URL" },
{ NULL }
}; };
static gint static gint

29
webrtc/sendrecv/gst/webrtc-sendrecv.c
View file

@ -50,6 +50,7 @@ static GOptionEntry entries[] =
{ {
{ "peer-id", 0, 0, G_OPTION_ARG_STRING, &peer_id, "String ID of the peer to connect to", "ID" }, { "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" }, { "server", 0, 0, G_OPTION_ARG_STRING, &server_url, "Signalling server to connect to", "URL" },
{ NULL },
}; };
static gboolean static gboolean
@ -105,12 +106,14 @@ handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name,
GstElement *q, *conv, *sink; GstElement *q, *conv, *sink;
GstPadLinkReturn ret; GstPadLinkReturn ret;
g_print ("Trying to handle stream with %s ! %s", convert_name, sink_name);
q = gst_element_factory_make ("queue", NULL); q = gst_element_factory_make ("queue", NULL);
g_assert (q); g_assert_nonnull (q);
conv = gst_element_factory_make (convert_name, NULL); conv = gst_element_factory_make (convert_name, NULL);
g_assert (conv); g_assert_nonnull (conv);
sink = gst_element_factory_make (sink_name, NULL); sink = gst_element_factory_make (sink_name, NULL);
g_assert (sink); g_assert_nonnull (sink);
gst_bin_add_many (GST_BIN (pipe), q, conv, sink, NULL); gst_bin_add_many (GST_BIN (pipe), q, conv, sink, NULL);
gst_element_sync_state_with_parent (q); gst_element_sync_state_with_parent (q);
gst_element_sync_state_with_parent (conv); gst_element_sync_state_with_parent (conv);
@ -120,7 +123,7 @@ handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name,
qpad = gst_element_get_static_pad (q, "sink"); qpad = gst_element_get_static_pad (q, "sink");
ret = gst_pad_link (pad, qpad); ret = gst_pad_link (pad, qpad);
g_assert (ret == GST_PAD_LINK_OK); g_assert_cmphex (ret, ==, GST_PAD_LINK_OK);
} }
static void static void
@ -224,9 +227,9 @@ on_offer_created (GstPromise * promise, gpointer user_data)
const GstStructure *reply; const GstStructure *reply;
gchar *desc; gchar *desc;
g_assert (app_state == PEER_CALL_NEGOTIATING); g_assert_cmphex (app_state, ==, PEER_CALL_NEGOTIATING);
g_assert (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); reply = gst_promise_get_reply (promise);
gst_structure_get (reply, "offer", gst_structure_get (reply, "offer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL); GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
@ -277,7 +280,7 @@ start_pipeline (void)
} }
webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "sendrecv"); webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "sendrecv");
g_assert (webrtc1 != NULL); g_assert_nonnull (webrtc1);
/* This is the gstwebrtc entry point where we create the offer and so on. It /* This is the gstwebrtc entry point where we create the offer and so on. It
* will be called when the pipeline goes to PLAYING. */ * will be called when the pipeline goes to PLAYING. */
@ -455,9 +458,9 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
GstSDPMessage *sdp; GstSDPMessage *sdp;
GstWebRTCSessionDescription *answer; GstWebRTCSessionDescription *answer;
g_assert (app_state == PEER_CALL_NEGOTIATING); g_assert_cmphex (app_state, ==, PEER_CALL_NEGOTIATING);
g_assert (json_object_has_member (object, "type")); g_assert_true (json_object_has_member (object, "type"));
/* In this example, we always create the offer and receive one answer. /* In this example, we always create the offer and receive one answer.
* See tests/examples/webrtcbidirectional.c in gst-plugins-bad for how to * See tests/examples/webrtcbidirectional.c in gst-plugins-bad for how to
* handle offers from peers and reply with answers using webrtcbin. */ * handle offers from peers and reply with answers using webrtcbin. */
@ -469,14 +472,14 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
g_print ("Received answer:\n%s\n", text); g_print ("Received answer:\n%s\n", text);
ret = gst_sdp_message_new (&sdp); ret = gst_sdp_message_new (&sdp);
g_assert (ret == GST_SDP_OK); g_assert_cmphex (ret, ==, GST_SDP_OK);
ret = gst_sdp_message_parse_buffer (text, strlen (text), sdp); ret = gst_sdp_message_parse_buffer (text, strlen (text), sdp);
g_assert (ret == GST_SDP_OK); g_assert_cmphex (ret, ==, GST_SDP_OK);
answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER, answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER,
sdp); sdp);
g_assert (answer); g_assert_nonnull (answer);
/* Set remote description on our pipeline */ /* Set remote description on our pipeline */
{ {
@ -523,7 +526,7 @@ on_server_connected (SoupSession * session, GAsyncResult * res,
return; return;
} }
g_assert (ws_conn != NULL); g_assert_nonnull (ws_conn);
app_state = SERVER_CONNECTED; app_state = SERVER_CONNECTED;
g_print ("Connected to signalling server\n"); g_print ("Connected to signalling server\n");