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rtspsrc: Fix critical while serializing timeout element message

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The "cause" field wasn't registered as a GEnumValue, so do that.

Fixes this critical in gst_structure_to_string():

`gst_value_serialize: assertion 'G_IS_VALUE (value)' failed`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1573>
This commit is contained in:
Nirbheek Chauhan 2022-01-26 11:22:31 +05:30 committed by GStreamer Marge Bot
parent 88e1b9081e
commit 980925a6a3
1 changed files with 20 additions and 2 deletions
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22
subprojects/gst-plugins-good/gst/rtsp/gstrtspsrc.c
View file

@ -161,6 +161,23 @@ enum _GstRtspSrcRtcpSyncMode
RTCP_SYNC_RTP
};
#define GST_TYPE_RTSP_SRC_TIMEOUT_CAUSE (gst_rtsp_src_timeout_cause_get_type())
static GType
gst_rtsp_src_timeout_cause_get_type (void)
{
static GType timeout_cause_type = 0;
static const GEnumValue timeout_causes[] = {
{GST_RTSP_SRC_TIMEOUT_CAUSE_RTCP, "timeout triggered by RTCP", "RTCP"},
{0, NULL, NULL},
};
if (!timeout_cause_type) {
timeout_cause_type =
g_enum_register_static ("GstRTSPSrcTimeoutCause", timeout_causes);
}
return timeout_cause_type;
}
enum _GstRtspSrcBufferMode
{
BUFFER_MODE_NONE,
@ -1267,6 +1284,7 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
gst_rtsp_ext_list_init ();
gst_type_mark_as_plugin_api (GST_TYPE_RTSP_SRC_TIMEOUT_CAUSE, 0);
gst_type_mark_as_plugin_api (GST_TYPE_RTSP_SRC_BUFFER_MODE, 0);
gst_type_mark_as_plugin_api (GST_TYPE_RTSP_SRC_NTP_TIME_SOURCE, 0);
gst_type_mark_as_plugin_api (GST_TYPE_RTSP_BACKCHANNEL, 0);
@ -3632,8 +3650,8 @@ on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
/* timeout, post element message */
gst_element_post_message (GST_ELEMENT_CAST (src),
gst_message_new_element (GST_OBJECT_CAST (src),
gst_structure_new ("GstRTSPSrcTimeout",
"cause", G_TYPE_ENUM, GST_RTSP_SRC_TIMEOUT_CAUSE_RTCP,
gst_structure_new ("GstRTSPSrcTimeout", "cause",
GST_TYPE_RTSP_SRC_TIMEOUT_CAUSE, GST_RTSP_SRC_TIMEOUT_CAUSE_RTCP,
"stream-number", G_TYPE_INT, stream->id, "ssrc", G_TYPE_UINT,
stream->ssrc, NULL)));