22 files changed, 131 insertions, 159 deletions
diff --git a/plugins/elements/gstcapsfilter.c b/plugins/elements/gstcapsfilter.c
index 30b6b99da..69b23f70c 100644
--- a/plugins/elements/gstcapsfilter.c
+++ b/plugins/elements/gstcapsfilter.c
@@ -21,19 +21,19 @@
  */
 /**
  * SECTION:element-capsfilter
+ * @title: capsfilter
  *
  * The element does not modify data as such, but can enforce limitations on the
  * data format.
  *
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
  * |[
  * gst-launch-1.0 videotestsrc ! capsfilter caps=video/x-raw,format=GRAY8 ! videoconvert ! autovideosink
  * ]| Limits acceptable video from videotestsrc to be grayscale. Equivalent to
  * |[
  * gst-launch-1.0 videotestsrc ! video/x-raw,format=GRAY8 ! videoconvert ! autovideosink
  * ]| which is a short notation for the capsfilter element.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/plugins/elements/gstconcat.c b/plugins/elements/gstconcat.c
index 788cc9c78..4cc134d99 100644
--- a/plugins/elements/gstconcat.c
+++ b/plugins/elements/gstconcat.c
@@ -20,6 +20,7 @@
  */
 /**
  * SECTION:element-concat
+ * @title: concat
  * @see_also: #GstFunnel
  *
  * Concatenates streams together to one continous stream.
@@ -37,12 +38,11 @@
  * another downstream element like a streamsynchronizer adjusts the base
  * values on its own). The adjust-base property can be used for this purpose.
  *
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
  * |[
  * gst-launch-1.0 concat name=c ! xvimagesink  videotestsrc num-buffers=100 ! c.   videotestsrc num-buffers=100 pattern=ball ! c.
  * ]| Plays two video streams one after another.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/plugins/elements/gstdataurisrc.c b/plugins/elements/gstdataurisrc.c
index d63e43728..e35d012dd 100644
--- a/plugins/elements/gstdataurisrc.c
+++ b/plugins/elements/gstdataurisrc.c
@@ -20,15 +20,17 @@
 
 /**
  * SECTION:element-dataurisrc
+ * @title: dataurisrc
  *
- * dataurisrc handles data: URIs, see <ulink url="http://tools.ietf.org/html/rfc2397">RFC 2397</ulink> for more information.
+ * dataurisrc handles data: URIs, see [RFC 2397](http://tools.ietf.org/html/rfc2397) for more information.
+ *
+ * ## Example launch line
  *
- * <refsect2>
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 -v dataurisrc uri="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAABAAAAAQCAYAAAAf8/9hAAAAfElEQVQ4je2MwQnAIAxFgziA4EnczIsO4MEROo/gzZWc4xdTbe1R6LGRR74heYS7iKElzfcMiRnt4hf8gk8EayB6luefue/HzlJfCA50XsNjYRxprZmenXNIKSGEsC+QUqK1hhgj521BzhnWWiilUGvdF5RS4L2HMQZCCJy8sHMm2TYdJAAAAABJRU5ErkJggg==" ! pngdec ! videoconvert ! imagefreeze ! videoconvert ! autovideosink
- * ]| This pipeline displays a small 16x16 PNG image from the data URI.
- * </refsect2>
+ * ]|
+ *
+ * This pipeline displays a small 16x16 PNG image from the data URI.
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/plugins/elements/gstdownloadbuffer.c b/plugins/elements/gstdownloadbuffer.c
index 9551d9bea..335dc5436 100644
--- a/plugins/elements/gstdownloadbuffer.c
+++ b/plugins/elements/gstdownloadbuffer.c
@@ -21,6 +21,7 @@
 
 /**
  * SECTION:element-downloadbuffer
+ * @title: downloadbuffer
  *
  * The downloadbuffer element provides on-disk buffering and caching of, typically,
  * a network file. temp-template should be set to a value such as
@@ -42,15 +43,12 @@
  * When the downloadbuffer has completely downloaded the media, it will
  * post an application message named  <classname>&quot;GstCacheDownloadComplete&quot;</classname>
  * with the following information:
- * <itemizedlist>
- * <listitem>
- *   <para>
+ *
+ * *
  *   G_TYPE_STRING
  *   <classname>&quot;location&quot;</classname>:
  *   the location of the completely downloaded file.
- *   </para>
- * </listitem>
- * </itemizedlist>
+ *
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/plugins/elements/gstfakesink.c b/plugins/elements/gstfakesink.c
index 0c7e84a55..b71f20f6f 100644
--- a/plugins/elements/gstfakesink.c
+++ b/plugins/elements/gstfakesink.c
@@ -21,16 +21,16 @@
  */
 /**
  * SECTION:element-fakesink
+ * @title: fakesink
  * @see_also: #GstFakeSrc
  *
  * Dummy sink that swallows everything.
- * 
- * <refsect2>
- * <title>Example launch line</title>
+ *
+ * ## Example launch line
  * |[
  * gst-launch-1.0 audiotestsrc num-buffers=1000 ! fakesink sync=false
  * ]| Render 1000 audio buffers (of default size) as fast as possible.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/plugins/elements/gstfakesrc.c b/plugins/elements/gstfakesrc.c
index 9d4041b6e..14f3226b0 100644
--- a/plugins/elements/gstfakesrc.c
+++ b/plugins/elements/gstfakesrc.c
@@ -21,6 +21,7 @@
  */
 /**
  * SECTION:element-fakesrc
+ * @title: fakesrc
  * @see_also: #GstFakeSink
  *
  * The fakesrc element is a multipurpose element that can generate
@@ -29,13 +30,12 @@
  * It is mostly used as a testing element, one trivial example for testing
  * basic <application>GStreamer</application> core functionality is:
  *
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
  * |[
  * gst-launch-1.0 -v fakesrc num-buffers=5 ! fakesink
  * ]| This pipeline will push 5 empty buffers to the fakesink element and then
  * sends an EOS.
- * </refsect2>
+ *
  */
 
 /* FIXME: this ignores basesrc::blocksize property, which could be used as an
diff --git a/plugins/elements/gstfakesrc.h b/plugins/elements/gstfakesrc.h
index 5190ee893..0d54d82cb 100644
--- a/plugins/elements/gstfakesrc.h
+++ b/plugins/elements/gstfakesrc.h
@@ -81,7 +81,7 @@ typedef enum {
 
 /**
  * GstFakeSrcFillType:
- * @FAKE_SRC_FILLTYPE_NOTHING: do not fill buffers 
+ * @FAKE_SRC_FILLTYPE_NOTHING: do not fill buffers
  * @FAKE_SRC_FILLTYPE_ZERO: fill buffers with 0
  * @FAKE_SRC_FILLTYPE_RANDOM: fill buffers with random bytes
  * @FAKE_SRC_FILLTYPE_PATTERN: fill buffers with a pattern
diff --git a/plugins/elements/gstfdsink.c b/plugins/elements/gstfdsink.c
index f4eb10073..ef2480087 100644
--- a/plugins/elements/gstfdsink.c
+++ b/plugins/elements/gstfdsink.c
@@ -22,6 +22,7 @@
 
 /**
  * SECTION:element-fdsink
+ * @title: fdsink
  * @see_also: #GstFdSrc
  *
  * Write data to a unix file descriptor.
diff --git a/plugins/elements/gstfdsrc.c b/plugins/elements/gstfdsrc.c
index 469a61423..963bcbd43 100644
--- a/plugins/elements/gstfdsrc.c
+++ b/plugins/elements/gstfdsrc.c
@@ -22,6 +22,7 @@
  */
 /**
  * SECTION:element-fdsrc
+ * @title: fdsrc
  * @see_also: #GstFdSink
  *
  * Read data from a unix file descriptor.
@@ -34,23 +35,16 @@
  * if no data was received in the given timeout.
  *
  * The message's structure contains one field:
- * <itemizedlist>
- * <listitem>
- *   <para>
- *   #guint64
- *   <classname>&quot;timeout&quot;</classname>: the timeout in microseconds that
+ *
+ * * #guint64 `timeout`: the timeout in microseconds that
  *   expired when waiting for data.
- *   </para>
- * </listitem>
- * </itemizedlist>
  *
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
  * |[
  * echo "Hello GStreamer" | gst-launch-1.0 -v fdsrc ! fakesink dump=true
  * ]| A simple pipeline to read from the standard input and dump the data
  * with a fakesink as hex ascii block.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/plugins/elements/gstfilesink.c b/plugins/elements/gstfilesink.c
index 9f0243bd7..4ce527e30 100644
--- a/plugins/elements/gstfilesink.c
+++ b/plugins/elements/gstfilesink.c
@@ -22,16 +22,16 @@
  */
 /**
  * SECTION:element-filesink
+ * @title: filesink
  * @see_also: #GstFileSrc
  *
  * Write incoming data to a file in the local file system.
  *
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
  * |[
  * gst-launch-1.0 v4l2src num-buffers=1 ! jpegenc ! filesink location=capture1.jpeg
  * ]| Capture one frame from a v4l2 camera and save as jpeg image.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/plugins/elements/gstfilesrc.c b/plugins/elements/gstfilesrc.c
index 15dd5a96b..530735196 100644
--- a/plugins/elements/gstfilesrc.c
+++ b/plugins/elements/gstfilesrc.c
@@ -21,16 +21,16 @@
  */
 /**
  * SECTION:element-filesrc
+ * @title: filesrc
  * @see_also: #GstFileSrc
  *
  * Read data from a file in the local file system.
  *
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
  * |[
  * gst-launch-1.0 filesrc location=song.ogg ! decodebin ! audioconvert ! audioresample ! autoaudiosink
  * ]| Play song.ogg audio file which must be in the current working directory.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H
@@ -648,7 +648,7 @@ gst_file_src_uri_set_uri (GstURIHandler * handler, const gchar * uri,
 #ifdef G_OS_WIN32
   /* Unfortunately, g_filename_from_uri() doesn't handle some UNC paths
    * correctly on windows, it leaves them with an extra backslash
-   * at the start if they're of the mozilla-style file://///host/path/file 
+   * at the start if they're of the mozilla-style file://///host/path/file
    * form. Correct this.
    */
   if (location[0] == '\\' && location[1] == '\\' && location[2] == '\\')
diff --git a/plugins/elements/gstfunnel.c b/plugins/elements/gstfunnel.c
index 3597d58f1..a9293383a 100644
--- a/plugins/elements/gstfunnel.c
+++ b/plugins/elements/gstfunnel.c
@@ -24,6 +24,7 @@
 
 /**
  * SECTION:element-funnel
+ * @title: funnel
  *
  * Takes packets from various input sinks into one output source.
  *
diff --git a/plugins/elements/gstidentity.c b/plugins/elements/gstidentity.c
index 77cd2d8e1..119fb5886 100644
--- a/plugins/elements/gstidentity.c
+++ b/plugins/elements/gstidentity.c
@@ -22,6 +22,7 @@
  */
 /**
  * SECTION:element-identity
+ * @title: identity
  *
  * Dummy element that passes incoming data through unmodified. It has some
  * useful diagnostic functions, such as offset and timestamp checking.
diff --git a/plugins/elements/gstinputselector.c b/plugins/elements/gstinputselector.c
index 5d1d65ecd..d842c2979 100644
--- a/plugins/elements/gstinputselector.c
+++ b/plugins/elements/gstinputselector.c
@@ -25,6 +25,7 @@
 
 /**
  * SECTION:element-input-selector
+ * @title: input-selector
  * @see_also: #GstOutputSelector
  *
  * Direct one out of N input streams to the output pad.
@@ -32,21 +33,12 @@
  * The input pads are from a GstPad subclass and have additional
  * properties, which users may find useful, namely:
  *
- * <itemizedlist>
- * <listitem>
- * "running-time": Running time of stream on pad (#gint64)
- * </listitem>
- * <listitem>
- * "tags": The currently active tags on the pad (#GstTagList, boxed type)
- * </listitem>
- * <listitem>
- * "active": If the pad is currently active (#gboolean)
- * </listitem>
- * <listitem>
- * "always-ok" : Make an inactive pads return #GST_FLOW_OK instead of
+ * * "running-time": Running time of stream on pad (#gint64)
+ * * "tags": The currently active tags on the pad (#GstTagList, boxed type)
+ * * "active": If the pad is currently active (#gboolean)
+ * * "always-ok" : Make an inactive pads return #GST_FLOW_OK instead of
  * #GST_FLOW_NOT_LINKED
- * </listitem>
- * </itemizedlist>
+ *
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/plugins/elements/gstmultiqueue.c b/plugins/elements/gstmultiqueue.c
index fed846d01..a9dbdbdfd 100644
--- a/plugins/elements/gstmultiqueue.c
+++ b/plugins/elements/gstmultiqueue.c
@@ -24,94 +24,70 @@
 
 /**
  * SECTION:element-multiqueue
+ * @title: multiqueue
  * @see_also: #GstQueue
  *
- * <refsect2>
- * <para>
  * Multiqueue is similar to a normal #GstQueue with the following additional
  * features:
- * <orderedlist>
- * <listitem>
- *   <itemizedlist><title>Multiple streamhandling</title>
- *   <listitem><para>
- *     The element handles queueing data on more than one stream at once. To
- *     achieve such a feature it has request sink pads (sink&percnt;u) and
- *     'sometimes' src pads (src&percnt;u).
- *   </para><para>
- *     When requesting a given sinkpad with gst_element_request_pad(),
- *     the associated srcpad for that stream will be created.
- *     Example: requesting sink1 will generate src1.
- *   </para></listitem>
- *   </itemizedlist>
- * </listitem>
- * <listitem>
- *   <itemizedlist><title>Non-starvation on multiple streams</title>
- *   <listitem><para>
- *     If more than one stream is used with the element, the streams' queues
- *     will be dynamically grown (up to a limit), in order to ensure that no
- *     stream is risking data starvation. This guarantees that at any given
- *     time there are at least N bytes queued and available for each individual
- *     stream.
- *   </para><para>
- *     If an EOS event comes through a srcpad, the associated queue will be
- *     considered as 'not-empty' in the queue-size-growing algorithm.
- *   </para></listitem>
- *   </itemizedlist>
- * </listitem>
- * <listitem>
- *   <itemizedlist><title>Non-linked srcpads graceful handling</title>
- *   <listitem><para>
- *     In order to better support dynamic switching between streams, the multiqueue
- *     (unlike the current GStreamer queue) continues to push buffers on non-linked
- *     pads rather than shutting down.
- *   </para><para>
- *     In addition, to prevent a non-linked stream from very quickly consuming all
- *     available buffers and thus 'racing ahead' of the other streams, the element
- *     must ensure that buffers and inlined events for a non-linked stream are pushed
- *     in the same order as they were received, relative to the other streams
- *     controlled by the element. This means that a buffer cannot be pushed to a
- *     non-linked pad any sooner than buffers in any other stream which were received
- *     before it.
- *   </para></listitem>
- *   </itemizedlist>
- * </listitem>
- * </orderedlist>
- * </para>
- * <para>
- *   Data is queued until one of the limits specified by the
- *   #GstMultiQueue:max-size-buffers, #GstMultiQueue:max-size-bytes and/or
- *   #GstMultiQueue:max-size-time properties has been reached. Any attempt to push
- *   more buffers into the queue will block the pushing thread until more space
- *   becomes available. #GstMultiQueue:extra-size-buffers,
- * </para>
- * <para>
- *   #GstMultiQueue:extra-size-bytes and #GstMultiQueue:extra-size-time are
- *   currently unused.
- * </para>
- * <para>
- *   The default queue size limits are 5 buffers, 10MB of data, or
- *   two second worth of data, whichever is reached first. Note that the number
- *   of buffers will dynamically grow depending on the fill level of 
- *   other queues.
- * </para>
- * <para>
- *   The #GstMultiQueue::underrun signal is emitted when all of the queues
- *   are empty. The #GstMultiQueue::overrun signal is emitted when one of the
- *   queues is filled.
- *   Both signals are emitted from the context of the streaming thread.
- * </para>
- * <para>
- *   When using #GstMultiQueue:sync-by-running-time the unlinked streams will
- *   be throttled by the highest running-time of linked streams. This allows
- *   further relinking of those unlinked streams without them being in the
- *   future (i.e. to achieve gapless playback).
- *   When dealing with streams which have got different consumption requirements
- *   downstream (ex: video decoders which will consume more buffer (in time) than
- *   audio decoders), it is recommended to group streams of the same type
- *   by using the pad "group-id" property. This will further throttle streams
- *   in time within that group.
- * </para>
- * </refsect2>
+ *
+ * 1) Multiple streamhandling
+ *
+ *  * The element handles queueing data on more than one stream at once. To
+ * achieve such a feature it has request sink pads (sink%u) and
+ * 'sometimes' src pads (src%u). When requesting a given sinkpad with gst_element_request_pad(),
+ * the associated srcpad for that stream will be created.
+ * Example: requesting sink1 will generate src1.
+ *
+ * 2) Non-starvation on multiple stream
+ *
+ * * If more than one stream is used with the element, the streams' queues
+ * will be dynamically grown (up to a limit), in order to ensure that no
+ * stream is risking data starvation. This guarantees that at any given
+ * time there are at least N bytes queued and available for each individual
+ * stream. If an EOS event comes through a srcpad, the associated queue will be
+ * considered as 'not-empty' in the queue-size-growing algorithm.
+ *
+ * 3) Non-linked srcpads graceful handling
+ *
+ * * In order to better support dynamic switching between streams, the multiqueue
+ * (unlike the current GStreamer queue) continues to push buffers on non-linked
+ * pads rather than shutting down. In addition, to prevent a non-linked stream from very quickly consuming all
+ * available buffers and thus 'racing ahead' of the other streams, the element
+ * must ensure that buffers and inlined events for a non-linked stream are pushed
+ * in the same order as they were received, relative to the other streams
+ * controlled by the element. This means that a buffer cannot be pushed to a
+ * non-linked pad any sooner than buffers in any other stream which were received
+ * before it.
+ *
+ * Data is queued until one of the limits specified by the
+ * #GstMultiQueue:max-size-buffers, #GstMultiQueue:max-size-bytes and/or
+ * #GstMultiQueue:max-size-time properties has been reached. Any attempt to push
+ * more buffers into the queue will block the pushing thread until more space
+ * becomes available. #GstMultiQueue:extra-size-buffers,
+ *
+ *
+ * #GstMultiQueue:extra-size-bytes and #GstMultiQueue:extra-size-time are
+ * currently unused.
+ *
+ * The default queue size limits are 5 buffers, 10MB of data, or
+ * two second worth of data, whichever is reached first. Note that the number
+ * of buffers will dynamically grow depending on the fill level of
+ * other queues.
+ *
+ * The #GstMultiQueue::underrun signal is emitted when all of the queues
+ * are empty. The #GstMultiQueue::overrun signal is emitted when one of the
+ * queues is filled.
+ * Both signals are emitted from the context of the streaming thread.
+ *
+ * When using #GstMultiQueue:sync-by-running-time the unlinked streams will
+ * be throttled by the highest running-time of linked streams. This allows
+ * further relinking of those unlinked streams without them being in the
+ * future (i.e. to achieve gapless playback).
+ * When dealing with streams which have got different consumption requirements
+ * downstream (ex: video decoders which will consume more buffer (in time) than
+ * audio decoders), it is recommended to group streams of the same type
+ * by using the pad "group-id" property. This will further throttle streams
+ * in time within that group.
  */
 
 #ifdef HAVE_CONFIG_H
@@ -504,7 +480,7 @@ gst_multi_queue_class_init (GstMultiQueueClass * klass)
    * size) is higher than the boundary values which can be set through the
    * GObject properties.
    *
-   * This can be used as an indicator of pre-roll. 
+   * This can be used as an indicator of pre-roll.
    */
   gst_multi_queue_signals[SIGNAL_OVERRUN] =
       g_signal_new ("overrun", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_FIRST,
@@ -551,8 +527,8 @@ gst_multi_queue_class_init (GstMultiQueueClass * klass)
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
   /**
-   * GstMultiQueue:use-buffering
-   * 
+   * GstMultiQueue:use-buffering:
+   *
    * Enable the buffering option in multiqueue so that BUFFERING messages are
    * emitted based on low-/high-percent thresholds.
    */
@@ -562,8 +538,8 @@ gst_multi_queue_class_init (GstMultiQueueClass * klass)
           DEFAULT_USE_BUFFERING, G_PARAM_READWRITE | GST_PARAM_MUTABLE_PLAYING |
           G_PARAM_STATIC_STRINGS));
   /**
-   * GstMultiQueue:low-percent
-   * 
+   * GstMultiQueue:low-percent:
+   *
    * Low threshold percent for buffering to start.
    */
   g_object_class_install_property (gobject_class, PROP_LOW_PERCENT,
@@ -573,8 +549,8 @@ gst_multi_queue_class_init (GstMultiQueueClass * klass)
           0, 100, DEFAULT_LOW_WATERMARK * 100,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
   /**
-   * GstMultiQueue:high-percent
-   * 
+   * GstMultiQueue:high-percent:
+   *
    * High threshold percent for buffering to finish.
    */
   g_object_class_install_property (gobject_class, PROP_HIGH_PERCENT,
@@ -584,7 +560,7 @@ gst_multi_queue_class_init (GstMultiQueueClass * klass)
           0, 100, DEFAULT_HIGH_WATERMARK * 100,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
   /**
-   * GstMultiQueue:low-watermark
+   * GstMultiQueue:low-watermark:
    *
    * Low threshold watermark for buffering to start.
    *
@@ -596,7 +572,7 @@ gst_multi_queue_class_init (GstMultiQueueClass * klass)
           0.0, 1.0, DEFAULT_LOW_WATERMARK,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
   /**
-   * GstMultiQueue:high-watermark
+   * GstMultiQueue:high-watermark:
    *
    * High threshold watermark for buffering to finish.
    *
@@ -609,8 +585,8 @@ gst_multi_queue_class_init (GstMultiQueueClass * klass)
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
   /**
-   * GstMultiQueue:sync-by-running-time
-   * 
+   * GstMultiQueue:sync-by-running-time:
+   *
    * If enabled multiqueue will synchronize deactivated or not-linked streams
    * to the activated and linked streams by taking the running time.
    * Otherwise multiqueue will synchronize the deactivated or not-linked
@@ -1334,7 +1310,7 @@ beach:
 
 
 /* calculate the diff between running time on the sink and src of the queue.
- * This is the total amount of time in the queue. 
+ * This is the total amount of time in the queue.
  * WITH LOCK TAKEN */
 static void
 update_time_level (GstMultiQueue * mq, GstSingleQueue * sq)
@@ -1464,7 +1440,7 @@ apply_buffer (GstMultiQueue * mq, GstSingleQueue * sq, GstClockTime timestamp,
 {
   GST_MULTI_QUEUE_MUTEX_LOCK (mq);
 
-  /* if no timestamp is set, assume it's continuous with the previous 
+  /* if no timestamp is set, assume it's continuous with the previous
    * time */
   if (timestamp == GST_CLOCK_TIME_NONE)
     timestamp = segment->position;
@@ -2580,7 +2556,7 @@ compute_high_id (GstMultiQueue * mq)
         lowest = sq->nextid;
     } else if (!GST_PAD_IS_EOS (sq->srcpad) && sq->srcresult != GST_FLOW_EOS) {
       /* If we don't have a global highid, or the global highid is lower than
-       * this single queue's last outputted id, store the queue's one, 
+       * this single queue's last outputted id, store the queue's one,
        * unless the singlequeue output is at EOS */
       if ((highid == G_MAXUINT32) || (sq->oldid > highid))
         highid = sq->oldid;
diff --git a/plugins/elements/gstoutputselector.c b/plugins/elements/gstoutputselector.c
index 15d547807..03cf20621 100644
--- a/plugins/elements/gstoutputselector.c
+++ b/plugins/elements/gstoutputselector.c
@@ -19,6 +19,7 @@
 
 /**
  * SECTION:element-output-selector
+ * @title: output-selector
  * @see_also: #GstOutputSelector, #GstInputSelector
  *
  * Direct input stream to one out of N output pads.
diff --git a/plugins/elements/gstqueue.c b/plugins/elements/gstqueue.c
index 57c57591e..e98c1f601 100644
--- a/plugins/elements/gstqueue.c
+++ b/plugins/elements/gstqueue.c
@@ -24,6 +24,7 @@
 
 /**
  * SECTION:element-queue
+ * @title: queue
  *
  * Data is queued until one of the limits specified by the
  * #GstQueue:max-size-buffers, #GstQueue:max-size-bytes and/or
diff --git a/plugins/elements/gstqueue2.c b/plugins/elements/gstqueue2.c
index 3a878e0d1..51db8285b 100644
--- a/plugins/elements/gstqueue2.c
+++ b/plugins/elements/gstqueue2.c
@@ -25,6 +25,7 @@
 
 /**
  * SECTION:element-queue2
+ * @title: queue2
  *
  * Data is queued until one of the limits specified by the
  * #GstQueue2:max-size-buffers, #GstQueue2:max-size-bytes and/or
diff --git a/plugins/elements/gststreamiddemux.c b/plugins/elements/gststreamiddemux.c
index 49fc29054..f0b1dd29b 100644
--- a/plugins/elements/gststreamiddemux.c
+++ b/plugins/elements/gststreamiddemux.c
@@ -24,6 +24,7 @@
 
 /**
  * SECTION:element-streamiddemux
+ * @title: streamiddemux
  *
  * The basic concept was started from de-funneling element which restores one
  * serialized stream via #GstFunnel to its original state. #GstStreamidDemux
diff --git a/plugins/elements/gsttee.c b/plugins/elements/gsttee.c
index 2369814bd..8531992cb 100644
--- a/plugins/elements/gsttee.c
+++ b/plugins/elements/gsttee.c
@@ -23,6 +23,7 @@
 
 /**
  * SECTION:element-tee
+ * @title: tee
  * @see_also: #GstIdentity
  *
  * Split data to multiple pads. Branching the data flow is useful when e.g.
@@ -34,14 +35,14 @@
  * provide separate threads for each branch. Otherwise a blocked dataflow in one
  * branch would stall the other branches.
  *
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
  * |[
  * gst-launch-1.0 filesrc location=song.ogg ! decodebin ! tee name=t ! queue ! audioconvert ! audioresample ! autoaudiosink t. ! queue ! audioconvert ! goom ! videoconvert ! autovideosink
- * ]| Play song.ogg audio file which must be in the current working directory
+ * ]|
+ *
+ * Play song.ogg audio file which must be in the current working directory
  * and render visualisations using the goom element (this can be easier done
  * using the playbin element, this is just an example pipeline).
- * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/plugins/elements/gsttypefindelement.c b/plugins/elements/gsttypefindelement.c
index fa192ca2a..5da2d7571 100644
--- a/plugins/elements/gsttypefindelement.c
+++ b/plugins/elements/gsttypefindelement.c
@@ -20,6 +20,7 @@
  */
 /**
  * SECTION:element-typefind
+ * @title: typefind
  *
  * Determines the media-type of a stream. It applies typefind functions in the
  * order of their rank. Once the type has been detected it sets its src pad caps
diff --git a/plugins/elements/gstvalve.c b/plugins/elements/gstvalve.c
index 04cc965b8..9dbc7a6f6 100644
--- a/plugins/elements/gstvalve.c
+++ b/plugins/elements/gstvalve.c
@@ -22,6 +22,7 @@
 
 /**
  * SECTION:element-valve
+ * @title: valve
  *
  * The valve is a simple element that drops buffers when the #GstValve:drop
  * property is set to %TRUE and lets then through otherwise.