diff --git a/ext/chromaprint/gstchromaprint.c b/ext/chromaprint/gstchromaprint.c
index 6ebadd31a4..bdfc8d1526 100644
--- a/ext/chromaprint/gstchromaprint.c
+++ b/ext/chromaprint/gstchromaprint.c
@@ -27,8 +27,8 @@
  *
  * The chromaprint element calculates an acoustic fingerprint for an
  * audio stream which can be used to identify a song and look up
- * further metadata from the <ulink url="http://acoustid.org/">Acoustid</ulink>
- * and Musicbrainz databases.
+ * further metadata from the [Acoustid](http://acoustid.org/) and Musicbrainz
+ * databases.
  *
  * ## Example launch line
  * |[
diff --git a/ext/directfb/dfbvideosink.c b/ext/directfb/dfbvideosink.c
index 2313e9bd6b..2e44872c22 100644
--- a/ext/directfb/dfbvideosink.c
+++ b/ext/directfb/dfbvideosink.c
@@ -23,19 +23,21 @@
  * @title: dfbvideosink
  *
  * DfbVideoSink renders video frames using the
- * <ulink url="http://www.directfb.org/">DirectFB</ulink> library.
+ * [DirectFB](http://www.directfb.org/) library.
  * Rendering can happen in two different modes :
  *
  * * Standalone: this mode will take complete control of the monitor forcing
- *   <ulink url="http://www.directfb.org/">DirectFB</ulink> to fullscreen layout.
+ *   DirectFB to fullscreen layout.
+ *
  *   This is convenient to test using the  gst-launch-1.0 command line tool or
  *   other simple applications. It is possible to interrupt playback while
  *   being in this mode by pressing the Escape key.
  *   This mode handles navigation events for every input device supported by
- *   the <ulink url="http://www.directfb.org/">DirectFB</ulink> library, it will
- *   look for available video modes in the fb.modes file and try to switch
- *   the framebuffer video mode to the most suitable one. Depending on
- *   hardware acceleration capabilities the element will handle scaling or not.
+ *   the DirectFB library, it will look for available video modes in the fb.modes
+ *   file and try to switch the framebuffer video mode to the most suitable one.
+ *   Depending on hardware acceleration capabilities the element will handle
+ *   scaling or not.
+ *
  *   If no acceleration is available it will do clipping or centering of the
  *   video frames respecting the original aspect ratio.
  *
@@ -43,7 +45,8 @@
  *   #GstDfbVideoSink:surface provided by the
  *   application developer. This is a more advanced usage of the element and
  *   it is required to integrate video playback in existing
- *   <ulink url="http://www.directfb.org/">DirectFB</ulink> applications.
+ *   DirectFB applications.
+ *
  *   When using this mode the element just renders to the
  *   #GstDfbVideoSink:surface provided by the
  *   application, that means it won't handle navigation events and won't resize
diff --git a/ext/fluidsynth/gstfluiddec.c b/ext/fluidsynth/gstfluiddec.c
index c9514dba08..48b53ca191 100644
--- a/ext/fluidsynth/gstfluiddec.c
+++ b/ext/fluidsynth/gstfluiddec.c
@@ -25,7 +25,7 @@
  * @see_also: timidity, wildmidi
  *
  * This element renders midi-events as audio streams using
- * <ulink url="http://fluidsynth.sourceforge.net//">Fluidsynth</ulink>.
+ * [Fluidsynth](http://fluidsynth.sourceforge.net/).
  * It offers better sound quality compared to the timidity or wildmidi element.
  *
  * ## Example pipeline
diff --git a/ext/kate/gstkatedec.c b/ext/kate/gstkatedec.c
index 65b29bfdbe..a2cec0970c 100644
--- a/ext/kate/gstkatedec.c
+++ b/ext/kate/gstkatedec.c
@@ -48,8 +48,9 @@
  * @title: katedec
  * @see_also: oggdemux
  *
- * This element decodes Kate streams
- * <ulink url="http://libkate.googlecode.com/">Kate</ulink> is a free codec
+ * This element decodes Kate streams.
+ *
+ * [Kate](http://libkate.googlecode.com/) is a free codec
  * for text based data, such as subtitles. Any number of kate streams can be
  * embedded in an Ogg stream.
  *
diff --git a/ext/kate/gstkateenc.c b/ext/kate/gstkateenc.c
index a76800191a..d2cbeb0c73 100644
--- a/ext/kate/gstkateenc.c
+++ b/ext/kate/gstkateenc.c
@@ -49,10 +49,11 @@
  * @title: kateenc
  * @see_also: oggmux
  *
- * This element encodes Kate streams
- * <ulink url="http://libkate.googlecode.com/">Kate</ulink> is a free codec
- * for text based data, such as subtitles. Any number of kate streams can be
- * embedded in an Ogg stream.
+ * This element encodes Kate streams.
+ *
+ * [Kate](http://libkate.googlecode.com/) is a free codec for text based data,
+ * such as subtitles. Any number of kate streams can be embedded in an Ogg
+ * stream.
  *
  * libkate (see above url) is needed to build this plugin.
  *
diff --git a/ext/kate/gstkatetiger.c b/ext/kate/gstkatetiger.c
index 21970c9f6d..8e877e77fa 100644
--- a/ext/kate/gstkatetiger.c
+++ b/ext/kate/gstkatetiger.c
@@ -48,12 +48,12 @@
  * @title: tiger
  * @see_also: katedec
  *
- * This element decodes and renders Kate streams
- * <ulink url="http://libkate.googlecode.com/">Kate</ulink> is a free codec
- * for text based data, such as subtitles. Any number of kate streams can be
- * embedded in an Ogg stream.
+ * This element decodes and renders Kate streams.
+ * [Kate](http://libkate.googlecode.com/) is a free codec for text based data,
+ * such as subtitles. Any number of kate streams can be embedded in an Ogg
+ * stream.
  *
- * libkate (see above url) and <ulink url="http://libtiger.googlecode.com/">libtiger</ulink>
+ * libkate (see above url) and [libtiger](http://libtiger.googlecode.com/)
  * are needed to build this element.
  *
  * ## Example pipeline
diff --git a/ext/ladspa/gstladspa.c b/ext/ladspa/gstladspa.c
index 2acd70f72a..92a6838a03 100644
--- a/ext/ladspa/gstladspa.c
+++ b/ext/ladspa/gstladspa.c
@@ -27,7 +27,8 @@
  * @see_also: #GstAudioConvert #GstAudioResample, #GstAudioTestSrc, #GstAutoAudioSink
  *
  * The LADSPA (Linux Audio Developer's Simple Plugin API) element is a bridge
- * for plugins using the <ulink url="http://www.ladspa.org/">LADSPA</ulink> API.
+ * for plugins using the [LADSPA](http://www.ladspa.org/) API.
+ *
  * It scans all installed LADSPA plugins and registers them as gstreamer
  * elements. If available it can also parse LRDF files and use the metadata for
  * element classification. The functionality you get depends on the LADSPA plugins
diff --git a/ext/lv2/gstlv2.c b/ext/lv2/gstlv2.c
index 705c4d404a..60d502f41a 100644
--- a/ext/lv2/gstlv2.c
+++ b/ext/lv2/gstlv2.c
@@ -30,8 +30,8 @@
  * a successor to LADSPA (Linux Audio Developer's Simple Plugin API).
  *
  * The LV2 element is a bridge for plugins using the
- * <ulink url="http://www.lv2plug.in/">LV2</ulink> API.  It scans all
- * installed LV2 plugins and registers them as gstreamer elements.
+ * [LV2](http://www.lv2plug.in/) API.  It scans all installed LV2 plugins and
+ * registers them as gstreamer elements.
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/ext/modplug/gstmodplug.cc b/ext/modplug/gstmodplug.cc
index a106f8c7f8..30c5952ab2 100644
--- a/ext/modplug/gstmodplug.cc
+++ b/ext/modplug/gstmodplug.cc
@@ -28,8 +28,8 @@
 /**
  * SECTION:element-modplug
  * 
- * Modplug uses the <ulink url="http://modplug-xmms.sourceforge.net/">modplug</ulink>
- * library to decode tracked music in the MOD/S3M/XM/IT and related formats.
+ * Modplug uses the [modplug](http://modplug-xmms.sourceforge.net/) library to
+ * decode tracked music in the MOD/S3M/XM/IT and related formats.
  * 
  * ## Example pipeline
  *
diff --git a/ext/mpeg2enc/gstmpeg2enc.cc b/ext/mpeg2enc/gstmpeg2enc.cc
index a86804d7bd..9bec3ef077 100644
--- a/ext/mpeg2enc/gstmpeg2enc.cc
+++ b/ext/mpeg2enc/gstmpeg2enc.cc
@@ -25,9 +25,10 @@
  * @see_also: mpeg2dec
  *
  * This element encodes raw video into an MPEG-1/2 elementary stream using the
- * <ulink url="http://mjpeg.sourceforge.net/">mjpegtools</ulink> library.
+ * [mjpegtools](http://mjpeg.sourceforge.net/) library.
+ *
  * Documentation on MPEG encoding in general can be found in the 
- * <ulink url="https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776">MJPEG Howto</ulink>
+ * [MJPEG Howto](https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776)
  * and on the various available parameters in the documentation
  * of the mpeg2enc tool in particular, which shares options with this element.
  *
diff --git a/ext/mplex/gstmplex.cc b/ext/mplex/gstmplex.cc
index f78e5bd340..5df12d6d03 100644
--- a/ext/mplex/gstmplex.cc
+++ b/ext/mplex/gstmplex.cc
@@ -26,9 +26,9 @@
  *
  * This element is an audio/video multiplexer for MPEG-1/2 video streams
  * and (un)compressed audio streams such as AC3, MPEG layer I/II/III.
- * It is based on the <ulink url="http://mjpeg.sourceforge.net/">mjpegtools</ulink> library.
+ * It is based on the [mjpegtools](http://mjpeg.sourceforge.net/) library.
  * Documentation on creating MPEG videos in general can be found in the
- * <ulink url="https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776">MJPEG Howto</ulink>
+ * [MJPEG Howto](https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776)
  * and the man-page of the mplex tool documents the properties of this element,
  * which are shared with the mplex tool.
  *
diff --git a/ext/openmpt/gstopenmptdec.c b/ext/openmpt/gstopenmptdec.c
index bc8c04fd58..9c84113158 100644
--- a/ext/openmpt/gstopenmptdec.c
+++ b/ext/openmpt/gstopenmptdec.c
@@ -23,8 +23,8 @@
  * @see_also: #GstOpenMptDec
  *
  * openmpdec decodes module music formats, such as S3M, MOD, XM, IT.
- * It uses the <ulink url="https://lib.openmpt.org">OpenMPT library</ulink>
- * for this purpose. It can be autoplugged and therefore works with decodebin.
+ * It uses the [OpenMPT library](https://lib.openmpt.org) for this purpose.
+ * It can be autoplugged and therefore works with decodebin.
  *
  * ## Example launch line
  *
diff --git a/ext/srt/gstsrtsink.c b/ext/srt/gstsrtsink.c
index 8d499f2fbb..591f414730 100644
--- a/ext/srt/gstsrtsink.c
+++ b/ext/srt/gstsrtsink.c
@@ -23,7 +23,7 @@
  * SECTION:element-srtsink
  * @title: srtsink
  *
- * srtsink is a network sink that sends <ulink url="http://www.srtalliance.org/">SRT</ulink>
+ * srtsink is a network sink that sends [SRT](http://www.srtalliance.org/)
  * packets to the network.
  *
  * ## Examples</title>
diff --git a/ext/srt/gstsrtsrc.c b/ext/srt/gstsrtsrc.c
index cc52a60faa..59103fe39d 100644
--- a/ext/srt/gstsrtsrc.c
+++ b/ext/srt/gstsrtsrc.c
@@ -23,7 +23,7 @@
  * SECTION:element-srtsrc
  * @title: srtsrc
  *
- * srtsrc is a network source that reads <ulink url="http://www.srtalliance.org/">SRT</ulink>
+ * srtsrc is a network source that reads [SRT](http://www.srtalliance.org/)
  * packets from the network.
  *
  * ## Examples
diff --git a/ext/voaacenc/gstvoaacenc.c b/ext/voaacenc/gstvoaacenc.c
index 0580d27f00..91eacb8c9c 100644
--- a/ext/voaacenc/gstvoaacenc.c
+++ b/ext/voaacenc/gstvoaacenc.c
@@ -21,8 +21,9 @@
  * SECTION:element-voaacenc
  * @title: voaacenc
  *
- * AAC audio encoder based on vo-aacenc library
- * <ulink url="http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/">vo-aacenc library source file</ulink>.
+ * AAC audio encoder based on vo-aacenc library.
+ *
+ * [vo-aacenc library source file](http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/)
  *
  * ## Example launch line
  * |[
diff --git a/ext/voamrwbenc/gstvoamrwbenc.c b/ext/voamrwbenc/gstvoamrwbenc.c
index c5eae31d7f..dfb997f454 100644
--- a/ext/voamrwbenc/gstvoamrwbenc.c
+++ b/ext/voamrwbenc/gstvoamrwbenc.c
@@ -23,7 +23,7 @@
  * @see_also: #GstAmrWbDec, #GstAmrWbParse
  *
  * AMR wideband encoder based on the
- * <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>.
+ * [reference codec implementation](http://www.penguin.cz/~utx/amr).
  *
  * ## Example launch line
  * |[
diff --git a/ext/wayland/gstwaylandsink.c b/ext/wayland/gstwaylandsink.c
index 78dd294a01..bfc4eb2ebf 100644
--- a/ext/wayland/gstwaylandsink.c
+++ b/ext/wayland/gstwaylandsink.c
@@ -27,8 +27,9 @@
  *
  *  The waylandsink is creating its own window and render the decoded video frames to that.
  *  Setup the Wayland environment as described in
- *  <ulink url="http://wayland.freedesktop.org/building.html">Wayland</ulink> home page.
- *  The current implementaion is based on weston compositor.
+ *  [Wayland](http://wayland.freedesktop.org/building.html) home page.
+ *
+ *  The current implementation is based on weston compositor.
  *
  * ## Example pipelines
  * |[
diff --git a/ext/webrtc/webrtcdatachannel.c b/ext/webrtc/webrtcdatachannel.c
index a4d49db145..693e626cd7 100644
--- a/ext/webrtc/webrtcdatachannel.c
+++ b/ext/webrtc/webrtcdatachannel.c
@@ -23,7 +23,7 @@
  * @title: GstWebRTCDataChannel
  * @see_also: #GstWebRTCRTPTransceiver
  *
- * <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransport">http://w3c.github.io/webrtc-pc/#dom-rtcsctptransport</ulink>
+ * <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransport>
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/ext/wildmidi/gstwildmididec.c b/ext/wildmidi/gstwildmididec.c
index ea88f74294..60ae3b3dea 100644
--- a/ext/wildmidi/gstwildmididec.c
+++ b/ext/wildmidi/gstwildmididec.c
@@ -23,8 +23,9 @@
  * @see_also: #GstWildmidiDec
  *
  * wildmididec decodes MIDI files.
- * It uses <ulink url="https://www.mindwerks.net/projects/wildmidi/">WildMidi</ulink>
- * for this purpose. It can be autoplugged and therefore works with decodebin.
+ *
+ * It uses [WildMidi](https://www.mindwerks.net/projects/wildmidi/) for this
+ * purpose. It can be autoplugged and therefore works with decodebin.
  *
  * ## Example launch line
  *
diff --git a/gst-libs/gst/webrtc/dtlstransport.c b/gst-libs/gst/webrtc/dtlstransport.c
index c3b2d519d5..ea7671fb25 100644
--- a/gst-libs/gst/webrtc/dtlstransport.c
+++ b/gst-libs/gst/webrtc/dtlstransport.c
@@ -23,7 +23,7 @@
  * @title: GstWebRTCDTLSTransport
  * @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver, #GstWebRTCICETransport
  *
- * <ulink url="https://www.w3.org/TR/webrtc/#rtcdtlstransport">https://www.w3.org/TR/webrtc/#rtcdtlstransport</ulink>
+ * <https://www.w3.org/TR/webrtc/#rtcdtlstransport>
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/gst-libs/gst/webrtc/icetransport.c b/gst-libs/gst/webrtc/icetransport.c
index e6f44378fe..d7e77d90f2 100644
--- a/gst-libs/gst/webrtc/icetransport.c
+++ b/gst-libs/gst/webrtc/icetransport.c
@@ -23,7 +23,7 @@
  * @title: GstWebRTCICETransport
  * @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver, #GstWebRTCDTLSTransport
  *
- * <ulink url="https://www.w3.org/TR/webrtc/#rtcicetransport">https://www.w3.org/TR/webrtc/#rtcicetransport</ulink>
+ * <https://www.w3.org/TR/webrtc/#rtcicetransport>
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/gst-libs/gst/webrtc/rtcsessiondescription.c b/gst-libs/gst/webrtc/rtcsessiondescription.c
index af5cd1c0d7..abdf5ca920 100644
--- a/gst-libs/gst/webrtc/rtcsessiondescription.c
+++ b/gst-libs/gst/webrtc/rtcsessiondescription.c
@@ -22,7 +22,7 @@
  * @short_description: RTCSessionDescription object
  * @title: GstWebRTCSessionDescription
  *
- * <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink>
+ * <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class>
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/gst-libs/gst/webrtc/rtcsessiondescription.h b/gst-libs/gst/webrtc/rtcsessiondescription.h
index 375642e767..5308c549a1 100644
--- a/gst-libs/gst/webrtc/rtcsessiondescription.h
+++ b/gst-libs/gst/webrtc/rtcsessiondescription.h
@@ -38,7 +38,7 @@ GType gst_webrtc_session_description_get_type (void);
  * @type: the #GstWebRTCSDPType of the description
  * @sdp: the #GstSDPMessage of the description
  *
- * See <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink>
+ * See <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class>
  */
 struct _GstWebRTCSessionDescription
 {
diff --git a/gst-libs/gst/webrtc/rtpreceiver.c b/gst-libs/gst/webrtc/rtpreceiver.c
index f21d77ef13..768e9876d3 100644
--- a/gst-libs/gst/webrtc/rtpreceiver.c
+++ b/gst-libs/gst/webrtc/rtpreceiver.c
@@ -23,7 +23,7 @@
  * @title: GstWebRTCRTPReceiver
  * @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPTransceiver
  *
- * <ulink url="https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface">https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface</ulink>
+ * <https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface>
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/gst-libs/gst/webrtc/rtpsender.c b/gst-libs/gst/webrtc/rtpsender.c
index da743f32d1..3a8a9044f0 100644
--- a/gst-libs/gst/webrtc/rtpsender.c
+++ b/gst-libs/gst/webrtc/rtpsender.c
@@ -23,7 +23,7 @@
  * @title: GstWebRTCRTPSender
  * @see_also: #GstWebRTCRTPReceiver, #GstWebRTCRTPTransceiver
  *
- * <ulink url="https://www.w3.org/TR/webrtc/#rtcrtpsender-interface">https://www.w3.org/TR/webrtc/#rtcrtpsender-interface</ulink>
+ * <https://www.w3.org/TR/webrtc/#rtcrtpsender-interface>
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/gst-libs/gst/webrtc/rtptransceiver.c b/gst-libs/gst/webrtc/rtptransceiver.c
index 8ea85f1686..08019462ad 100644
--- a/gst-libs/gst/webrtc/rtptransceiver.c
+++ b/gst-libs/gst/webrtc/rtptransceiver.c
@@ -23,7 +23,7 @@
  * @title: GstWebRTCRTPTransceiver
  * @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver
  *
- * <ulink url="https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface">https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface</ulink>
+ * <https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface>
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/gst-libs/gst/webrtc/webrtc_fwd.h b/gst-libs/gst/webrtc/webrtc_fwd.h
index 07d9b39ec2..61c1aca9e5 100644
--- a/gst-libs/gst/webrtc/webrtc_fwd.h
+++ b/gst-libs/gst/webrtc/webrtc_fwd.h
@@ -82,7 +82,7 @@ typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/
  * @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>
+ * See <http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate>
  */
 typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/
 {
@@ -101,7 +101,7 @@ typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/
  * @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>
+ * See <http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate>
  */
 typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/
 {
@@ -123,7 +123,7 @@ typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/
  * @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>
+ * See <http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate>
  */
 typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/
 {
@@ -144,7 +144,7 @@ typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/
  * @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>
+ * See <http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate>
  */
 typedef enum /*< underscore_name=gst_webrtc_peer_connection_state >*/
 {
@@ -185,7 +185,7 @@ typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
  * @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>
+ * See <http://w3c.github.io/webrtc-pc/#rtcsdptype>
  */
 typedef enum /*< underscore_name=gst_webrtc_sdp_type >*/
 {
@@ -282,7 +282,7 @@ typedef enum /*< underscore_name=gst_webrtc_fec_type >*/
  * GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected
  * GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
  *
- * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate">http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate</ulink>
+ * See <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate>
  *
  * Since: 1.16
  */
@@ -301,7 +301,7 @@ typedef enum /*< underscore_name=gst_webrtc_sctp_transport_state >*/
  * GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium
  * GST_WEBRTC_PRIORITY_TYPE_HIGH: high
  *
- * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype">http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype</ulink>
+ * See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
  *
  * Since: 1.16
  */
@@ -321,7 +321,7 @@ typedef enum /*< underscore_name=gst_webrtc_priority_type >*/
  * GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing
  * GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
  *
- * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate">http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate</ulink>
+ * See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate>
  *
  * Since: 1.16
  */
diff --git a/gst/accurip/gstaccurip.c b/gst/accurip/gstaccurip.c
index 05578d4e18..a3d4877342 100644
--- a/gst/accurip/gstaccurip.c
+++ b/gst/accurip/gstaccurip.c
@@ -38,8 +38,8 @@
  *
  * The accurip element calculates a CRC for an audio stream which can be used
  * to match the audio stream to a database hosted on
- * <ulink url="http://accuraterip.com/">AccurateRip</ulink>. This database
- * is used to check for a CD rip accuracy.
+ * [AccurateRip](http://accuraterip.com/). This database is used to check for a
+ * CD rip accuracy.
  *
  * ## Example launch line
  * |[
diff --git a/gst/festival/gstfestival.c b/gst/festival/gstfestival.c
index e3206ef768..fbe9ecdae9 100644
--- a/gst/festival/gstfestival.c
+++ b/gst/festival/gstfestival.c
@@ -65,9 +65,9 @@
  * @title: festival
  *
  * This element connects to a
- * <ulink url="http://www.festvox.org/festival/index.html">festival</ulink>
- * server process and uses it to synthesize speech. Festival need to run already
- * in server mode, started as `festival --server`
+ * [festival](http://www.festvox.org/festival/index.html) server process and
+ * uses it to synthesize speech. Festival need to run already in server mode,
+ * started as `festival --server`
  *
  * ## Example pipeline
  * |[
diff --git a/gst/pcapparse/gstpcapparse.c b/gst/pcapparse/gstpcapparse.c
index 8a60cdad6c..aab2b8a1f5 100644
--- a/gst/pcapparse/gstpcapparse.c
+++ b/gst/pcapparse/gstpcapparse.c
@@ -26,9 +26,8 @@
  * #GstPcapParse:src-port and #GstPcapParse:dst-port to restrict which packets
  * should be included.
  *
- * The supported data format is the classical <ulink
- * url="https://wiki.wireshark.org/Development/LibpcapFileFormat">libpcap file
- * format</ulink>.
+ * The supported data format is the classical
+ * [libpcap file format](https://wiki.wireshark.org/Development/LibpcapFileFormat)
  *
  * ## Example pipelines
  * |[