webrtc: Document more objects

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>

diff --git a/gst-libs/gst/webrtc/rtpreceiver.h b/gst-libs/gst/webrtc/rtpreceiver.h
index 55a9a86..746bd4f 100644 (file)
--- 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 0c5c077..b521448 100644 (file)
--- a/gst-libs/gst/webrtc/rtpsender.h
+++ b/gst-libs/gst/webrtc/rtpsender.h
@@ -41,6 +41,12 @@ 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 5d14e95..6a7564c 100644 (file)
--- a/gst-libs/gst/webrtc/rtptransceiver.h
+++ b/gst-libs/gst/webrtc/rtptransceiver.h
@@ -38,7 +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
 {