From 78c687da3e351493fad017d93bfc2a1c0b47425d Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= <olivier.crete@collabora.com>
Date: Fri, 2 Oct 2020 21:38:00 -0400
Subject: [PATCH] webrtc: Document more objects

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
---
 gst-libs/gst/webrtc/rtpreceiver.h    |  8 ++++++++
 gst-libs/gst/webrtc/rtpsender.h      | 10 ++++++++++
 gst-libs/gst/webrtc/rtptransceiver.h | 22 ++++++++++++++++++++++
 3 files changed, 40 insertions(+)

diff --git a/gst-libs/gst/webrtc/rtpreceiver.h b/gst-libs/gst/webrtc/rtpreceiver.h
index 55a9a86fd9..746bd4fae7 100644
--- a/gst-libs/gst/webrtc/rtpreceiver.h
+++ b/gst-libs/gst/webrtc/rtpreceiver.h
@@ -37,6 +37,14 @@ 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 bcaf93c604..ca7296b265 100644
--- a/gst-libs/gst/webrtc/rtpsender.h
+++ b/gst-libs/gst/webrtc/rtpsender.h
@@ -37,6 +37,16 @@ GType gst_webrtc_rtp_sender_get_type(void);
 
 /**
  * GstWebRTCRTPSender:
+ * @transport: The transport for RTP packets
+ * @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 4b2e6e30cc..73391f06b0 100644
--- a/gst-libs/gst/webrtc/rtptransceiver.h
+++ b/gst-libs/gst/webrtc/rtptransceiver.h
@@ -38,6 +38,28 @@ 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
 {