Gst.WebRTCStatsType – gstreamer-webrtc-1.0 Reference Manual
WebRTCStatsType
Description:
[
CCode
( cprefix =
"GST_WEBRTC_STATS_"
, type_id =
"gst_webrtc_stats_type_get_type ()"
) ]
public
enum
WebRTCStatsType
Namespace:
Gst
Package:
gstreamer-webrtc-1.0
Content:
Enum values:
CANDIDATE_PAIR
-
candidate-pair
CERTIFICATE
-
certificate
CODEC
-
codec
CSRC
-
csrc
DATA_CHANNEL
-
data-channel
INBOUND_RTP
-
inbound-rtp
LOCAL_CANDIDATE
-
local-candidate
OUTBOUND_RTP
-
outbound-rtp
PEER_CONNECTION
-
peer-connectiion
REMOTE_CANDIDATE
-
remote-candidate
REMOTE_INBOUND_RTP
-
remote-inbound-rtp
REMOTE_OUTBOUND_RTP
-
remote-outbound-rtp
STREAM
-
stream
TRANSPORT
-
transport
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