diff options
| author | Nirbheek Chauhan <nirbheek@centricular.com> | 2022-01-26 11:22:31 +0530 |
|---|---|---|
| committer | Tim-Philipp Müller <tim@centricular.com> | 2022-01-28 10:18:02 +0000 |
| commit | ed327fd38f8072ee82cb95dac727cd5ccfc7a9a7 (patch) | |
| tree | 88fa7866f77295ea5b12eb30b5d2cfe379e172f8 | |
| parent | f2638a486a80f22cff3e8bcaf07f90b4e7138fe4 (diff) | |
| download | gstreamer-plugins-good-ed327fd38f8072ee82cb95dac727cd5ccfc7a9a7.tar.gz | |
rtspsrc: Fix critical while serializing timeout element message
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/gst-plugins-good/-/merge_requests/1111>
| -rw-r--r-- | gst/rtsp/gstrtspsrc.c | 22 |
1 files changed, 20 insertions, 2 deletions
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 2180d175e..35405bd9b 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -160,6 +160,23 @@ 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, @@ -1225,6 +1242,7 @@ 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); @@ -3584,8 +3602,8 @@ 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))); |