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webrtc: Improve GstWebRTCStatsType docstring
Fix a typo of peer-connectiion -> peer-connection Add a link to the w3c RTCStats type for a description of what each statistics type is. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3434>
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1 changed files with 3 additions and 1 deletions
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@ -308,7 +308,7 @@ typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/
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* @GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
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* @GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
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* @GST_WEBRTC_STATS_CSRC: csrc
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* @GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
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* @GST_WEBRTC_STATS_PEER_CONNECTION: peer-connection
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* @GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
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* @GST_WEBRTC_STATS_STREAM: stream
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* @GST_WEBRTC_STATS_TRANSPORT: transport
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@ -316,6 +316,8 @@ typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/
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* @GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
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* @GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
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* @GST_WEBRTC_STATS_CERTIFICATE: certificate
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*
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* See <https://w3c.github.io/webrtc-stats/#dom-rtcstatstype>
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*/
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typedef enum /*< underscore_name=gst_webrtc_stats_type >*/
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{
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