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| author | Marijn Suijten <marijns95@gmail.com> | 2021-03-28 12:03:09 +0200 |
|---|---|---|
| committer | GStreamer Marge Bot <gitlab-merge-bot@gstreamer-foundation.org> | 2021-03-28 13:08:24 +0000 |
| commit | 061e32b197d7cedb72618cbe552008f36da94055 (patch) | |
| tree | fd543aa10c442569acf8e8a29dc683359321bf8e /gst-libs | |
| parent | 0b916e7cec578f56f96e8c18476003093621db44 (diff) | |
| download | gstreamer-plugins-bad-061e32b197d7cedb72618cbe552008f36da94055.tar.gz | |
Add @ prefix to enum-variant references in documentation
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>
Diffstat (limited to 'gst-libs')
| -rw-r--r-- | gst-libs/gst/d3d11/gstd3d11memory.h | 8 | ||||
| -rw-r--r-- | gst-libs/gst/webrtc/webrtc_fwd.h | 38 |
2 files changed, 23 insertions, 23 deletions
diff --git a/gst-libs/gst/d3d11/gstd3d11memory.h b/gst-libs/gst/d3d11/gstd3d11memory.h index 23382edae..abfb0cd6a 100644 --- a/gst-libs/gst/d3d11/gstd3d11memory.h +++ b/gst-libs/gst/d3d11/gstd3d11memory.h @@ -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 */ diff --git a/gst-libs/gst/webrtc/webrtc_fwd.h b/gst-libs/gst/webrtc/webrtc_fwd.h index 4de341231..87d5f98d5 100644 --- a/gst-libs/gst/webrtc/webrtc_fwd.h +++ b/gst-libs/gst/webrtc/webrtc_fwd.h @@ -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. |