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-rw-r--r--docs/plugins/farsight2-plugins-sections.txt7
-rw-r--r--gst/fsrtpconference/fs-rtp-dtmf-event-source.c2
-rw-r--r--gst/fsrtpconference/fs-rtp-dtmf-sound-source.c2
-rw-r--r--gst/fsrtpconference/fs-rtp-session.c2
-rw-r--r--gst/fsrtpconference/fs-rtp-substream.c2
5 files changed, 11 insertions, 4 deletions
diff --git a/docs/plugins/farsight2-plugins-sections.txt b/docs/plugins/farsight2-plugins-sections.txt
index 71610bf7..97f031c7 100644
--- a/docs/plugins/farsight2-plugins-sections.txt
+++ b/docs/plugins/farsight2-plugins-sections.txt
@@ -507,3 +507,10 @@ fs_raw_participant_get_type
<SUBSECTION Private>
fs_raw_participant_new
</SECTION>
+
+
+<SECTION>
+<FILE>element-fsrtcpfilter</FILE>
+<TITLE>FsRtcpFilter</TITLE>
+FsRtcpFilter
+</SECTION>
diff --git a/gst/fsrtpconference/fs-rtp-dtmf-event-source.c b/gst/fsrtpconference/fs-rtp-dtmf-event-source.c
index 01476f7d..6962c199 100644
--- a/gst/fsrtpconference/fs-rtp-dtmf-event-source.c
+++ b/gst/fsrtpconference/fs-rtp-dtmf-event-source.c
@@ -37,7 +37,7 @@
#define GST_CAT_DEFAULT fsrtpconference_debug
-/**
+/*
* SECTION:fs-rtp-dtmf-event-source
* @short_description: Class to create the source of DTMF events
*
diff --git a/gst/fsrtpconference/fs-rtp-dtmf-sound-source.c b/gst/fsrtpconference/fs-rtp-dtmf-sound-source.c
index a359c81e..d3c0667b 100644
--- a/gst/fsrtpconference/fs-rtp-dtmf-sound-source.c
+++ b/gst/fsrtpconference/fs-rtp-dtmf-sound-source.c
@@ -38,7 +38,7 @@
#define GST_CAT_DEFAULT fsrtpconference_debug
-/**
+/*
* SECTION:fs-rtp-dtmf-sound-source
* @short_description: Class to create the source of DTMF sounds
*
diff --git a/gst/fsrtpconference/fs-rtp-session.c b/gst/fsrtpconference/fs-rtp-session.c
index a5024ac0..70548c41 100644
--- a/gst/fsrtpconference/fs-rtp-session.c
+++ b/gst/fsrtpconference/fs-rtp-session.c
@@ -3195,7 +3195,7 @@ link_other_pads (gpointer item, GValue *ret, gpointer user_data)
return FALSE;
}
-/**
+/*
* @codec: The currently selected codec for sending (but not the send_codec)
*/
diff --git a/gst/fsrtpconference/fs-rtp-substream.c b/gst/fsrtpconference/fs-rtp-substream.c
index 22ce131d..afd89da9 100644
--- a/gst/fsrtpconference/fs-rtp-substream.c
+++ b/gst/fsrtpconference/fs-rtp-substream.c
@@ -38,7 +38,7 @@
#define GST_CAT_DEFAULT fsrtpconference_debug
-/**
+/*
* SECTION:fs-rtp-sub-stream
* @short_description: The receive codec bin for a ssrc and a pt
*