Minor documentation fixes

ext/closedcaption/io-sim.c

@@ -47,7 +47,7 @@
 
 #include "io-sim.h"
 
-/**
+/*
  * @addtogroup Rawenc Raw VBI encoder
  * @ingroup Raw
  * @brief Converting sliced VBI data to raw VBI images.
@@ -375,7 +375,7 @@ clear_image (uint8_t * p,
   }
 }
 
-/**
+/*
  * @param raw Noise will be added to this raw VBI image.
  * @param sp Describes the raw VBI data in the buffer. @a sp->sampling_format
  *   must be @c VBI_PIXFMT_Y8 (@c VBI_PIXFMT_YUV420 in libzvbi 0.2.x).
@@ -1008,12 +1008,12 @@ _vbi_raw_video_image (uint8_t * raw,
   return TRUE;
 }
 
-/**
+/*
  * @example examples/rawout.c
  * Raw VBI output example.
  */
 
-/**
+/*
  * @param raw A raw VBI image will be stored here.
  * @param raw_size Size of the @a raw buffer in bytes. The buffer
  *   must be large enough for @a sp->count[0] + count[1] lines
@@ -1079,7 +1079,7 @@ vbi_raw_vbi_image (uint8_t * raw,
       swap_fields ? _VBI_RAW_SWAP_FIELDS : 0, sliced, n_sliced_lines);
 }
 
-/**
+/*
  * @param raw A raw VBI image will be stored here.
  * @param raw_size Size of the @a raw buffer in bytes. The buffer
  *   must be large enough for @a sp->count[0] + count[1] lines

ext/curl/gstcurlbasesink.c

@@ -21,7 +21,6 @@
  * SECTION:element-curlsink
  * @title: curlsink
  * @short_description: sink that uploads data to a server using libcurl
- * @see_also:
  *
  * This is a network sink that uses libcurl as a client to upload data to
  * a server (e.g. a HTTP/FTP server).

ext/curl/gstcurlfilesink.c

@@ -21,7 +21,6 @@
  * SECTION:element-curlfilesink
  * @title: curlfilesink
  * @short_description: sink that uploads data to a server using libcurl
- * @see_also:
  *
  * This is a network sink that uses libcurl as a client to upload data to
  * a local or network drive.

ext/curl/gstcurlftpsink.c

@@ -21,7 +21,6 @@
  * SECTION:element-curlftpsink
  * @title: curlftpsink
  * @short_description: sink that uploads data to a server using libcurl
- * @see_also:
  *
  * This is a network sink that uses libcurl as a client to upload data to
  * an FTP server.

ext/curl/gstcurlhttpsink.c

@@ -21,7 +21,6 @@
  * SECTION:element-curlhttpsink
  * @title: curlhttpsink
  * @short_description: sink that uploads data to a server using libcurl
- * @see_also:
  *
  * This is a network sink that uses libcurl as a client to upload data to
  * an HTTP server.

ext/curl/gstcurlsftpsink.c

@@ -21,7 +21,6 @@
  * SECTION:element-curlsftpsink
  * @title: curlsftpsink
  * @short_description: sink that uploads data to a server using libcurl
- * @see_also:
  *
  * This is a network sink that uses libcurl as a client to upload data to
  * a SFTP (SSH File Transfer Protocol) server.

ext/curl/gstcurlsmtpsink.c

@@ -21,7 +21,6 @@
  * SECTION:element-curlsink
  * @title: curlsink
  * @short_description: sink that uploads data to a server using libcurl
- * @see_also:
  *
  * This is a network sink that uses libcurl as a client to upload data to
  * an SMTP server.

ext/curl/gstcurlsshsink.c

@@ -21,7 +21,6 @@
  * SECTION:element-curlsshsink
  * @title: curlsshsink
  * @short_description: sink that uploads data to a server using libcurl
- * @see_also:
  *
  * This is a network sink that uses libcurl.
  *

ext/curl/gstcurltlssink.c

@@ -21,7 +21,6 @@
  * SECTION:element-curltlssink
  * @title: curltlssink
  * @short_description: sink that uploads data to a server using libcurl
- * @see_also:
  *
  * This is a network sink that uses libcurl.
  *

ext/smoothstreaming/gstmssmanifest.c

@@ -975,7 +975,7 @@ gst_mss_manifest_get_duration (GstMssManifest * manifest)
 }
 
 
-/**
+/*
  * Gets the duration in nanoseconds
  */
 GstClockTime
@@ -1194,7 +1194,7 @@ gst_mss_stream_type_name (GstMssStreamType streamtype)
   }
 }
 
-/**
+/*
  * Seeks all streams to the fragment that contains the set time
  *
  * @forward: if this is forward playback
@@ -1215,7 +1215,7 @@ gst_mss_manifest_seek (GstMssManifest * manifest, gboolean forward,
     ((forward && (flags & GST_SEEK_FLAG_SNAP_AFTER)) || \
     (!forward && (flags & GST_SEEK_FLAG_SNAP_BEFORE)))
 
-/**
+/*
  * Seeks this stream to the fragment that contains the sample at time
  *
  * @time: time in nanoseconds

ext/webrtc/gstwebrtcbin.c

@@ -1188,7 +1188,7 @@ _check_if_negotiation_is_needed (GstWebRTCBin * webrtc)
   /* FIXME: emit when input caps/format changes? */
 
   /* If connection has created any RTCDataChannel's, and no m= section has
-   * been negotiated yet for data, return "true". 
+   * been negotiated yet for data, return "true".
    * FIXME */
 
   if (!webrtc->current_local_description) {
@@ -3674,7 +3674,7 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
 
   if (sd->sdp->type == GST_WEBRTC_SDP_TYPE_ROLLBACK) {
     /* FIXME:
-     * If the mid value of an RTCRtpTransceiver was set to a non-null value 
+     * If the mid value of an RTCRtpTransceiver was set to a non-null value
      * by the RTCSessionDescription that is being rolled back, set the mid
      * value of that transceiver to null, as described by [JSEP]
      * (section 4.1.7.2.).
@@ -5054,7 +5054,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::create-offer:
-   * @object: the #GstWebRtcBin
+   * @object: the #webrtcbin
    * @options: create-offer options
    * @promise: a #GstPromise which will contain the offer
    */
@@ -5067,7 +5067,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::create-answer:
-   * @object: the #GstWebRtcBin
+   * @object: the #webrtcbin
    * @options: create-answer options
    * @promise: a #GstPromise which will contain the answer
    */
@@ -5080,7 +5080,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::set-local-description:
-   * @object: the #GstWebRtcBin
+   * @object: the #GstWebRTCBin
    * @desc: a #GstWebRTCSessionDescription description
    * @promise: (nullable): a #GstPromise to be notified when it's set
    */
@@ -5093,7 +5093,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::set-remote-description:
-   * @object: the #GstWebRtcBin
+   * @object: the #GstWebRTCBin
    * @desc: a #GstWebRTCSessionDescription description
    * @promise: (nullable): a #GstPromise to be notified when it's set
    */
@@ -5106,7 +5106,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::add-ice-candidate:
-   * @object: the #GstWebRtcBin
+   * @object: the #webrtcbin
    * @mline_index: the index of the media description in the SDP
    * @ice-candidate: an ice candidate
    */
@@ -5118,7 +5118,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::get-stats:
-   * @object: the #GstWebRtcBin
+   * @object: the #webrtcbin
    * @pad: (nullable): A #GstPad to get the stats for, or %NULL for all
    * @promise: a #GstPromise for the result
    *
@@ -5195,7 +5195,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::on-negotiation-needed:
-   * @object: the #GstWebRtcBin
+   * @object: the #webrtcbin
    */
   gst_webrtc_bin_signals[ON_NEGOTIATION_NEEDED_SIGNAL] =
       g_signal_new ("on-negotiation-needed", G_TYPE_FROM_CLASS (klass),
@@ -5204,7 +5204,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::on-ice-candidate:
-   * @object: the #GstWebRtcBin
+   * @object: the #webrtcbin
    * @mline_index: the index of the media description in the SDP
    * @candidate: the ICE candidate
    */
@@ -5215,7 +5215,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::on-new-transceiver:
-   * @object: the #GstWebRtcBin
+   * @object: the #webrtcbin
    * @candidate: the new #GstWebRTCRTPTransceiver
    */
   gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL] =
@@ -5225,8 +5225,8 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::on-data-channel:
-   * @object: the #GstWebRtcBin
-   * @candidate: the new #GstWebRTCDataChannel
+   * @object: the #GstWebRTCBin
+   * @candidate: the new `GstWebRTCDataChannel`
    */
   gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL] =
       g_signal_new ("on-data-channel", G_TYPE_FROM_CLASS (klass),
@@ -5235,7 +5235,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::add-transceiver:
-   * @object: the #GstWebRtcBin
+   * @object: the #webrtcbin
    * @direction: the direction of the new transceiver
    * @caps: (allow none): the codec preferences for this transceiver
    *
@@ -5250,7 +5250,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::get-transceivers:
-   * @object: the #GstWebRtcBin
+   * @object: the #webrtcbin
    *
    * Returns: a #GArray of #GstWebRTCRTPTransceivers
    */
@@ -5262,7 +5262,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::get-transceiver:
-   * @object: the #GstWebRtcBin
+   * @object: the #GstWebRTCBin
    * @idx: The index of the transceiver
    *
    * Returns: the #GstWebRTCRTPTransceiver, or %NULL
@@ -5277,7 +5277,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /**
    * GstWebRTCBin::add-turn-server:
-   * @object: the #GstWebRtcBin
+   * @object: the #GstWebRTCBin
    * @uri: The uri of the server of the form turn(s)://username:password@host:port
    *
    * Add a turn server to obtain ICE candidates from
@@ -5290,7 +5290,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
 
   /*
    * GstWebRTCBin::create-data-channel:
-   * @object: the #GstWebRtcBin
+   * @object: the #GstWebRTCBin
    * @label: the label for the data channel
    * @options: a #GstStructure of options for creating the data channel
    *

ext/webrtc/gstwebrtcice.c

@@ -941,7 +941,7 @@ gst_webrtc_ice_class_init (GstWebRTCICEClass * klass)
 
   /**
    * GstWebRTCICE::on-ice-candidate:
-   * @object: the #GstWebRtcBin
+   * @object: the #GstWebRTCBin
    * @candidate: the ICE candidate
    */
   gst_webrtc_ice_signals[ON_ICE_CANDIDATE_SIGNAL] =

gst-libs/gst/adaptivedemux/gstadaptivedemux.c

@@ -22,7 +22,6 @@
 /**
  * SECTION:gstadaptivedemux
  * @short_description: Base class for adaptive demuxers
- * @see_also:
  *
  * What is an adaptive demuxer?
  * Adaptive demuxers are special demuxers in the sense that they don't

gst-libs/gst/codecparsers/gsth264parser.h

@@ -658,7 +658,7 @@ struct _GstH264SPSExtMVCLevelValue
  *   level values signalled for the coded video sequence.
  * @level_value: array of #GstH264SPSExtMVCLevelValue
  *
- * Represents the parsed seq_parameter_set_mvc_extension().
+ * Represents the parsed `seq_parameter_set_mvc_extension()`.
  *
  * Since: 1.6
 	 */

gst-libs/gst/codecparsers/gsth265parser.c

@@ -39,7 +39,7 @@
  *
  *   * From #GST_H265_NAL_SLICE_TRAIL_N to #GST_H265_NAL_SLICE_CRA_NUT: gst_h265_parser_parse_slice_hdr()
  *
- *   * #GST_H265_NAL_SEI: gst_h265_parser_parse_sei()
+ *   * `GST_H265_NAL_*_SEI`: gst_h265_parser_parse_sei()
  *
  *   * #GST_H265_NAL_VPS: gst_h265_parser_parse_vps()
  *
@@ -2119,7 +2119,7 @@ gst_h265_parser_parse_pps (GstH265Parser * parser,
 /**
  * gst_h265_parser_parse_slice_hdr:
  * @parser: a #GstH265Parser
- * @nalu: The #GST_H265_NAL_SLICE #GstH265NalUnit to parse
+ * @nalu: The `GST_H265_NAL_SLICE` #GstH265NalUnit to parse
  * @slice: The #GstH265SliceHdr to fill.
  *
  * Parses @data, and fills the @slice structure.
@@ -2656,7 +2656,7 @@ gst_h265_sei_free (GstH265SEIMessage * sei)
 /**
  * gst_h265_parser_parse_sei:
  * @nalparser: a #GstH265Parser
- * @nalu: The #GST_H265_NAL_SEI #GstH265NalUnit to parse
+ * @nalu: The `GST_H265_NAL_*_SEI` #GstH265NalUnit to parse
  * @messages: The GArray of #GstH265SEIMessage to fill. The caller must free it when done.
  *
  * Parses @data, create and fills the @messages array.

gst-libs/gst/codecparsers/nalutils.c

@@ -27,7 +27,7 @@
  * Boston, MA 02110-1301, USA.
  */
 
-/**
+/*
  * Common code for NAL parsing from h264 and h265 parsers.
  */
 

gst-libs/gst/mpegts/gst-dvb-descriptor.c

@@ -1988,7 +1988,7 @@ gst_mpegts_descriptor_parse_dvb_multilingual_component (const
  * @private_data_specifier: (out): the private data specifier id
  * registered by http://www.dvbservices.com/
  * @private_data: (out) (transfer full) (allow-none) (array length=length): additional data or NULL
- * @length: (out) (allow-none): length of %private_data
+ * @length: (out) (allow-none): length of @private_data
  *
  * Parses out the private data specifier from the @descriptor.
  *
@@ -2024,7 +2024,7 @@ gst_mpegts_descriptor_parse_dvb_private_data_specifier (const
  * @descriptor: a %GST_MTS_DESC_DVB_FREQUENCY_LIST #GstMpegtsDescriptor
  * @offset: (out): %FALSE in Hz, %TRUE in kHz
  * @list: (out) (transfer full) (element-type guint32): a list of all frequencies in Hz/kHz
- * depending on %offset
+ * depending on @offset
  *
  * Parses out a list of frequencies from the @descriptor.
  *
@@ -2195,7 +2195,7 @@ gst_mpegts_descriptor_parse_dvb_scrambling (const GstMpegtsDescriptor *
  * @descriptor: a %GST_MTS_DESC_DVB_DATA_BROADCAST_ID #GstMpegtsDescriptor
  * @data_broadcast_id: (out): the data broadcast id
  * @id_selector_bytes: (out) (transfer full) (array length=len): the selector bytes, if present
- * @len: (out): the length of #id_selector_bytes
+ * @len: (out): the length of @id_selector_bytes
  *
  * Parses out the data broadcast id from the @descriptor.
  *

gst-libs/gst/mpegts/gst-dvb-descriptor.h

@@ -464,8 +464,8 @@ struct _GstMpegtsDVBLinkageExtendedEvent
  * @transport_stream_id: the transport id
  * @original_network_id: the original network id
  * @service_id: the service id
- * @linkage_type: the type which %linkage_data has
- * @private_data_length: the length for %private_data_bytes
+ * @linkage_type: the type which @linkage_data has
+ * @private_data_length: the length for @private_data_bytes
  * @private_data_bytes: additional data bytes
  */
 struct _GstMpegtsDVBLinkageDescriptor

gst-libs/gst/player/gstplayer-g-main-context-signal-dispatcher.c

@@ -200,7 +200,7 @@ static void
  * @application_context: (allow-none): GMainContext to use or %NULL
  *
  * Creates a new GstPlayerSignalDispatcher that uses @application_context,
- * or the thread default one if %NULL is used. See gst_player_new_full().
+ * or the thread default one if %NULL is used. See gst_player_new().
  *
  * Returns: (transfer full): the new GstPlayerSignalDispatcher
  */

gst-libs/gst/player/gstplayer-media-info.c

@@ -784,7 +784,7 @@ gst_player_media_info_get_container_format (const GstPlayerMediaInfo * info)
  * @info: a #GstPlayerMediaInfo
  *
  * Function to get the image (or preview-image) stored in taglist.
- * Application can use gst_sample_*_() API's to get caps, buffer etc.
+ * Application can use `gst_sample_*_()` API's to get caps, buffer etc.
  *
  * Returns: (transfer none): GstSample or NULL.
  */

gst-libs/gst/player/gstplayer.c

@@ -69,6 +69,9 @@ GST_DEBUG_CATEGORY_STATIC (gst_player_debug);
 #define DEFAULT_AUDIO_VIDEO_OFFSET 0
 #define DEFAULT_SUBTITLE_VIDEO_OFFSET 0
 
+/**
+ * gst_player_error_quark:
+ */
 GQuark
 gst_player_error_quark (void)
 {

gst-libs/gst/webrtc/dtlstransport.h

@@ -35,6 +35,9 @@ GType gst_webrtc_dtls_transport_get_type(void);
 #define GST_IS_WEBRTC_DTLS_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DTLS_TRANSPORT))
 #define GST_WEBRTC_DTLS_TRANSPORT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DTLS_TRANSPORT,GstWebRTCDTLSTransportClass))
 
+/**
+ * GstWebRTCDTLSTransport:
+ */
 struct _GstWebRTCDTLSTransport
 {
   GstObject                          parent;

gst-libs/gst/webrtc/icetransport.h

@@ -34,6 +34,9 @@ GType gst_webrtc_ice_transport_get_type(void);
 #define GST_IS_WEBRTC_ICE_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ICE_TRANSPORT))
 #define GST_WEBRTC_ICE_TRANSPORT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ICE_TRANSPORT,GstWebRTCICETransportClass))
 
+/**
+ * GstWebRTCICETransport:
+ */
 struct _GstWebRTCICETransport
 {
   GstObject                          parent;

gst-libs/gst/webrtc/rtpreceiver.h

@@ -35,6 +35,9 @@ GType gst_webrtc_rtp_receiver_get_type(void);
 #define GST_IS_WEBRTC_RTP_RECEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_RECEIVER))
 #define GST_WEBRTC_RTP_RECEIVER_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_RECEIVER,GstWebRTCRTPReceiverClass))
 
+/**
+ * GstWebRTCRTPReceiver:
+ */
 struct _GstWebRTCRTPReceiver
 {
   GstObject                          parent;

gst-libs/gst/webrtc/rtpsender.h

@@ -35,6 +35,9 @@ GType gst_webrtc_rtp_sender_get_type(void);
 #define GST_IS_WEBRTC_RTP_SENDER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_SENDER))
 #define GST_WEBRTC_RTP_SENDER_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSenderClass))
 
+/**
+ * GstWebRTCRTPSender:
+ */
 struct _GstWebRTCRTPSender
 {
   GstObject                          parent;

gst-libs/gst/webrtc/rtptransceiver.h

@@ -36,6 +36,9 @@ GType gst_webrtc_rtp_transceiver_get_type(void);
 #define GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER))
 #define GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER,GstWebRTCRTPTransceiverClass))
 
+/**
+ * GstWebRTCRTPTransceiver:
+ */
 struct _GstWebRTCRTPTransceiver
 {
   GstObject                         parent;

gst-libs/gst/webrtc/webrtc_fwd.h

@@ -27,6 +27,11 @@
 
 #include <gst/gst.h>
 
+/**
+ * SECTION:webrtc_fwd.h
+ * @title: GstWebRTC Enumerations
+ */
+
 #ifndef GST_WEBRTC_API
 # ifdef BUILDING_GST_WEBRTC
 #  define GST_WEBRTC_API GST_API_EXPORT         /* from config.h */
@@ -56,11 +61,11 @@ typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass;
 
 /**
  * GstWebRTCDTLSTransportState:
- * GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
- * GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
- * GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
- * GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
- * GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected
+ * @GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
+ * @GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
+ * @GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
+ * @GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
+ * @GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected
  */
 typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/
 {
@@ -73,9 +78,9 @@ typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/
 
 /**
  * GstWebRTCICEGatheringState:
- * GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
- * GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
- * GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
+ * @GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
+ * @GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
+ * @GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
  *
  * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink>
  */
@@ -88,13 +93,13 @@ typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/
 
 /**
  * GstWebRTCICEConnectionState:
- * GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
- * GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
- * GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
- * GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
- * GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
- * GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
- * GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
+ * @GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
+ * @GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
+ * @GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
+ * @GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
+ * @GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
+ * @GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
+ * @GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
  *
  * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink>
  */
@@ -111,12 +116,12 @@ typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/
 
 /**
  * GstWebRTCSignalingState:
- * GST_WEBRTC_SIGNALING_STATE_STABLE: stable
- * GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
- * GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
- * GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
- * GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
- * GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
+ * @GST_WEBRTC_SIGNALING_STATE_STABLE: stable
+ * @GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
+ * @GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
+ * @GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
+ * @GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
+ * @GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
  *
  * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate">http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate</ulink>
  */
@@ -132,12 +137,12 @@ typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/
 
 /**
  * GstWebRTCPeerConnectionState:
- * GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
- * GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
- * GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
- * GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
- * GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
- * GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
+ * @GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
+ * @GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
+ * @GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
+ * @GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
+ * @GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
+ * @GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
  *
  * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink>
  */
@@ -153,8 +158,8 @@ typedef enum /*< underscore_name=gst_webrtc_peer_connection_state >*/
 
 /**
  * GstWebRTCICERole:
- * GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
- * GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
+ * @GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
+ * @GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
  */
 typedef enum /*< underscore_name=gst_webrtc_ice_role >*/
 {
@@ -164,8 +169,8 @@ typedef enum /*< underscore_name=gst_webrtc_ice_role >*/
 
 /**
  * GstWebRTCICEComponent:
- * GST_WEBRTC_ICE_COMPONENT_RTP,
- * GST_WEBRTC_ICE_COMPONENT_RTCP,
+ * @GST_WEBRTC_ICE_COMPONENT_RTP: RTP component
+ * @GST_WEBRTC_ICE_COMPONENT_RTCP: RTCP component
  */
 typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
 {
@@ -175,10 +180,10 @@ typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
 
 /**
  * GstWebRTCSDPType:
- * GST_WEBRTC_SDP_TYPE_OFFER: offer
- * GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
- * GST_WEBRTC_SDP_TYPE_ANSWER: answer
- * GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
+ * @GST_WEBRTC_SDP_TYPE_OFFER: offer
+ * @GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
+ * @GST_WEBRTC_SDP_TYPE_ANSWER: answer
+ * @GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
  *
  * See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink>
  */
@@ -191,12 +196,12 @@ typedef enum /*< underscore_name=gst_webrtc_sdp_type >*/
 } GstWebRTCSDPType;
 
 /**
- * GstWebRTCRtpTransceiverDirection:
- * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: none
- * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: inactive
- * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: sendonly
- * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: recvonly
- * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: sendrecv
+ * GstWebRTCRTPTransceiverDirection:
+ * @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: none
+ * @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: inactive
+ * @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: sendonly
+ * @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: recvonly
+ * @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: sendrecv
  */
 typedef enum /*< underscore_name=gst_webrtc_rtp_transceiver_direction >*/
 {
@@ -209,10 +214,10 @@ typedef enum /*< underscore_name=gst_webrtc_rtp_transceiver_direction >*/
 
 /**
  * GstWebRTCDTLSSetup:
- * GST_WEBRTC_DTLS_SETUP_NONE: none
- * GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
- * GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
- * GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
+ * @GST_WEBRTC_DTLS_SETUP_NONE: none
+ * @GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
+ * @GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
+ * @GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
  */
 typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/
 {
@@ -224,20 +229,20 @@ typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/
 
 /**
  * GstWebRTCStatsType:
- * GST_WEBRTC_STATS_CODEC: codec
- * GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
- * GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
- * GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
- * GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
- * GST_WEBRTC_STATS_CSRC: csrc
- * GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
- * GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
- * GST_WEBRTC_STATS_STREAM: stream
- * GST_WEBRTC_STATS_TRANSPORT: transport
- * GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
- * GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
- * GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
- * GST_WEBRTC_STATS_CERTIFICATE: certificate
+ * @GST_WEBRTC_STATS_CODEC: codec