webrtc: Document more objects

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
This commit is contained in:
Olivier Crête 2020-10-02 21:38:00 -04:00
parent 8ba08598bb
commit ad68c6b1eb
3 changed files with 35 additions and 0 deletions

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@ -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
{

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@ -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
{

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@ -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
{