mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 02:31:03 +00:00
webrtc: Document more objects
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
This commit is contained in:
parent
9a796482dc
commit
78c687da3e
3 changed files with 40 additions and 0 deletions
|
@ -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
|
||||
{
|
||||
|
|
|
@ -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
|
||||
{
|
||||
|
|
|
@ -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
|
||||
{
|
||||
|
|
Loading…
Reference in a new issue