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Add @ prefix to enum-variant references in documentation

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Found while working on GStreamer-rs documentation, some enums had this
bit of text pasted verbatim in the enum documentation rather than
attached to the enum-variant.  Fortunately it seems these in WebRTC and
D3D11 are the only ones matching the non-@-prefixed pattern:

    ^ \* GST_\w+:\s*\w+

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2118>
This commit is contained in:
Marijn Suijten 2021-03-28 12:03:09 +02:00 committed by GStreamer Marge Bot
parent 0b916e7cec
commit 061e32b197
2 changed files with 23 additions and 23 deletions
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8
gst-libs/gst/d3d11/gstd3d11memory.h
View file

@ -79,10 +79,10 @@ G_BEGIN_DECLS
/**
* GstD3D11AllocationFlags:
* GST_D3D11_ALLOCATION_FLAG_TEXTURE_ARRAY: Indicates each allocated texture
* should be array type. This type of
* is used for D3D11/DXVA decoders
* in general.
* @GST_D3D11_ALLOCATION_FLAG_TEXTURE_ARRAY: Indicates each allocated texture
* should be array type. This type of
* is used for D3D11/DXVA decoders
* in general.
*
* Since: 1.20
*/

38
gst-libs/gst/webrtc/webrtc_fwd.h
View file

@ -280,10 +280,10 @@ typedef enum /*< underscore_name=gst_webrtc_fec_type >*/
/**
* GstWebRTCSCTPTransportState:
* GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new
* GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting
* GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected
* GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
* @GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new
* @GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting
* @GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected
* @GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
*
* See <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate>
*
@ -299,10 +299,10 @@ typedef enum /*< underscore_name=gst_webrtc_sctp_transport_state >*/
/**
* GstWebRTCPriorityType:
* GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low
* GST_WEBRTC_PRIORITY_TYPE_LOW: low
* GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium
* GST_WEBRTC_PRIORITY_TYPE_HIGH: high
* @GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low
* @GST_WEBRTC_PRIORITY_TYPE_LOW: low
* @GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium
* @GST_WEBRTC_PRIORITY_TYPE_HIGH: high
*
* See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
*
@ -318,11 +318,11 @@ typedef enum /*< underscore_name=gst_webrtc_priority_type >*/
/**
* GstWebRTCDataChannelState:
* GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new
* GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection
* GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open
* GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing
* GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
* @GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new
* @GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection
* @GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open
* @GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing
* @GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
*
* See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate>
*
@ -339,10 +339,10 @@ typedef enum /*< underscore_name=gst_webrtc_data_channel_state >*/
/**
* GstWebRTCBundlePolicy:
* GST_WEBRTC_BUNDLE_POLICY_NONE: none
* GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced
* GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat
* GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
* @GST_WEBRTC_BUNDLE_POLICY_NONE: none
* @GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced
* @GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat
* @GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
*
* See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
* for more information.
@ -359,8 +359,8 @@ typedef enum /*<underscore_name=gst_webrtc_bundle_policy>*/
/**
* GstWebRTCICETransportPolicy:
* GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all
* GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
* @GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all
* @GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
*
* See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
* for more information.