diff --git a/gir-files/Gst-1.0.gir b/gir-files/Gst-1.0.gir
index 7a32293ca..3a47f3459 100644
--- a/gir-files/Gst-1.0.gir
+++ b/gir-files/Gst-1.0.gir
@@ -44717,11 +44717,11 @@ determine a order for the two provided values.
The major version of GStreamer at compile time:
-
+
The micro version of GStreamer at compile time:
-
+
The minor version of GStreamer at compile time:
diff --git a/gir-files/GstAudio-1.0.gir b/gir-files/GstAudio-1.0.gir
index 2187c9dc4..e35d0857c 100644
--- a/gir-files/GstAudio-1.0.gir
+++ b/gir-files/GstAudio-1.0.gir
@@ -8417,6 +8417,7 @@ functionality.
diff --git a/gir-files/GstPbutils-1.0.gir b/gir-files/GstPbutils-1.0.gir
index 596a61bed..57b7c6ca9 100644
--- a/gir-files/GstPbutils-1.0.gir
+++ b/gir-files/GstPbutils-1.0.gir
@@ -2744,13 +2744,13 @@ in debugging.
The micro version of GStreamer's gst-plugins-base libraries at compile time.
The minor version of GStreamer's gst-plugins-base libraries at compile time.
diff --git a/gir-files/GstWebRTC-1.0.gir b/gir-files/GstWebRTC-1.0.gir
index f5e002e83..7ab709f09 100644
--- a/gir-files/GstWebRTC-1.0.gir
+++ b/gir-files/GstWebRTC-1.0.gir
@@ -15,24 +15,33 @@ and/or use gtk-doc annotations. -->
shared-library="libgstwebrtc-1.0.so.0"
c:identifier-prefixes="Gst"
c:symbol-prefixes="gst">
-
+
GST_WEBRTC_DTLS_SETUP_NONE: none
GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
-
+
+ c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS"
+ glib:nick="actpass">
+ c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE"
+ glib:nick="active">
+ c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE"
+ glib:nick="passive">
transfer-ownership="none">
-
-
+
+
@@ -140,6 +149,8 @@ GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
@@ -148,36 +159,50 @@ GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected
+ c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW"
+ glib:nick="new">
+ c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED"
+ glib:nick="closed">
+ c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED"
+ glib:nick="failed">
+ c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING"
+ glib:nick="connecting">
+ c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED"
+ glib:nick="connected">
-
+
GST_WEBRTC_ICE_COMPONENT_RTP,
GST_WEBRTC_ICE_COMPONENT_RTCP,
-
+
+ c:identifier="GST_WEBRTC_ICE_COMPONENT_RTCP"
+ glib:nick="rtcp">
GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
@@ -189,34 +214,43 @@ 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>
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW"
+ glib:nick="new">
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING"
+ glib:nick="checking">
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED"
+ glib:nick="connected">
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED"
+ glib:nick="completed">
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED"
+ glib:nick="failed">
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED"
+ glib:nick="disconnected">
+ c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED"
+ glib:nick="closed">
GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
@@ -224,27 +258,35 @@ 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>
+ c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_NEW"
+ glib:nick="new">
+ c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING"
+ glib:nick="gathering">
+ c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE"
+ glib:nick="complete">
-
+
GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
+ c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED"
+ glib:nick="controlled">
+ c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING"
+ glib:nick="controlling">
-
+
-
-
+
+
-
-
+
+
@@ -410,6 +449,8 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
@@ -420,27 +461,33 @@ 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>
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_NEW"
+ glib:nick="new">
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING"
+ glib:nick="connecting">
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED"
+ glib:nick="connected">
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED"
+ glib:nick="disconnected">
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED"
+ glib:nick="failed">
+ c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED"
+ glib:nick="closed">
+ c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE"
+ glib:nick="none">
+ c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE"
+ glib:nick="inactive">
+ c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY"
+ glib:nick="sendonly">
+ c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY"
+ glib:nick="recvonly">
+ c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV"
+ glib:nick="sendrecv">
-
+
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>
-
+
+ c:identifier="GST_WEBRTC_SDP_TYPE_PRANSWER"
+ glib:nick="pranswer">
+ c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER"
+ glib:nick="answer">
+ c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK"
+ glib:nick="rollback">
+
+
+ the string representation of @type or "unknown" when @type is not
+ recognized.
+
+
+
+
+ a #GstWebRTCSDPType
+
+
+
+
-
+
GST_WEBRTC_SIGNALING_STATE_STABLE: stable
GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
@@ -769,30 +848,39 @@ 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>
+ c:identifier="GST_WEBRTC_SIGNALING_STATE_STABLE"
+ glib:nick="stable">
+ c:identifier="GST_WEBRTC_SIGNALING_STATE_CLOSED"
+ glib:nick="closed">
+ c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER"
+ glib:nick="have-local-offer">
+ c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER"
+ glib:nick="have-remote-offer">
+ c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER"
+ glib:nick="have-local-pranswer">
+ c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER"
+ glib:nick="have-remote-pranswer">
-
+
GST_WEBRTC_STATS_CODEC: codec
GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
@@ -807,59 +895,80 @@ 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
-
+
+ c:identifier="GST_WEBRTC_STATS_INBOUND_RTP"
+ glib:nick="inbound-rtp">
+ c:identifier="GST_WEBRTC_STATS_OUTBOUND_RTP"
+ glib:nick="outbound-rtp">
+ c:identifier="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP"
+ glib:nick="remote-inbound-rtp">
+ c:identifier="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP"
+ glib:nick="remote-outbound-rtp">
-
+
+ c:identifier="GST_WEBRTC_STATS_PEER_CONNECTION"
+ glib:nick="peer-connection">
+ c:identifier="GST_WEBRTC_STATS_DATA_CHANNEL"
+ glib:nick="data-channel">
-
+
+ c:identifier="GST_WEBRTC_STATS_TRANSPORT"
+ glib:nick="transport">
+ c:identifier="GST_WEBRTC_STATS_CANDIDATE_PAIR"
+ glib:nick="candidate-pair">
+ c:identifier="GST_WEBRTC_STATS_LOCAL_CANDIDATE"
+ glib:nick="local-candidate">
+ c:identifier="GST_WEBRTC_STATS_REMOTE_CANDIDATE"
+ glib:nick="remote-candidate">
+ c:identifier="GST_WEBRTC_STATS_CERTIFICATE"
+ glib:nick="certificate">
+ c:identifier="gst_webrtc_sdp_type_to_string"
+ moved-to="WebRTCSDPType.to_string">
the string representation of @type or "unknown" when @type is not
recognized.