Revert "webrtc: Document more objects"

This reverts commit ad68c6b1eb7c73c66dc9d1dbf1a8cc47fd489c61.

It breaks the CI until the C# bindings are fixed.

diff --git a/gst-libs/gst/webrtc/rtpreceiver.h b/gst-libs/gst/webrtc/rtpreceiver.h
index 746bd4fae7e0262d59a442f68de1d0b2b26afe39..55a9a86fd9b6bc9d03b88dc379512ac981c2a006 100644 (file)
--- a/gst-libs/gst/webrtc/rtpreceiver.h
+++ b/gst-libs/gst/webrtc/rtpreceiver.h
@@ -37,14 +37,6 @@ GType gst_webrtc_rtp_receiver_get_type(void);
 
 /**
  * GstWebRTCRTPReceiver:
- * @transport: The transport for RTP packets
- * @rtcp_transport: The transport for RTCP packets without rtcp-mux
- *
- * An object to track the receiving aspect of the stream
- *
- * Mostly matches the WebRTC RTCRtpReceiver interface.
- *
- * Since: 1.16
  */
 struct _GstWebRTCRTPReceiver
 {

diff --git a/gst-libs/gst/webrtc/rtpsender.h b/gst-libs/gst/webrtc/rtpsender.h
index b521448f62487712afd7554f4c30b560dee0b559..0c5c07751aeae9577ae3fd1555f0e2c4d49b7e9d 100644 (file)
--- a/gst-libs/gst/webrtc/rtpsender.h
+++ b/gst-libs/gst/webrtc/rtpsender.h
@@ -41,12 +41,6 @@ GType gst_webrtc_rtp_sender_get_type(void);
  * @rtcp_transport: The transport for RTCP packets without rtcp-mux
  * @send_encodings: Unused
  * @priority: The priority of the stream (Since: 1.20)
- *
- * An object to track the sending aspect of the stream
- *
- * Mostly matches the WebRTC RTCRtpSender interface.
- *
- * Since: 1.16
  */
 struct _GstWebRTCRTPSender
 {

diff --git a/gst-libs/gst/webrtc/rtptransceiver.h b/gst-libs/gst/webrtc/rtptransceiver.h
index 6a7564c8c27f19c6c0c8b1652924006ef1bec8bf..5d14e95f1ee156c1a0c85598f58723ac42dfe443 100644 (file)
--- a/gst-libs/gst/webrtc/rtptransceiver.h
+++ b/gst-libs/gst/webrtc/rtptransceiver.h
@@ -38,28 +38,7 @@ GType gst_webrtc_rtp_transceiver_get_type(void);
 
 /**
  * GstWebRTCRTPTransceiver:
- * @mline: the mline number this transceiver corresponds to
- * @mid: The media ID of the m-line associated with this
- * transceiver. This association is established, when possible,
- * whenever either a local or remote description is applied. This
- * field is NULL if neither a local or remote description has been
- * applied, or if its associated m-line is rejected by either a remote
- * offer or any answer.
- * @stopped: Indicates whether or not sending and receiving using the paired
- * #GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled,
- * either due to SDP offer/answer
- * @sender: The #GstWebRTCRTPSender object responsible sending  data to the
- * remote peer
- * @receiver: The #GstWebRTCRTPReceiver object responsible for receiver data from
- * the remote peer.
- * @direction: The transceiver's desired direction.
- * @current_direction: The transceiver's current direction (read-only)
- * @codec_preferences: A caps representing the codec preferences (read-only)
  * @kind: Type of media (Since: 1.20)
- *
- * Mostly matches the WebRTC RTCRtpTransceiver interface.
- *
- * Since: 1.16
  */
 struct _GstWebRTCRTPTransceiver
 {