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| author | Matthew Waters <matthew@centricular.com> | 2018-09-10 23:52:05 +1000 |
|---|---|---|
| committer | Matthew Waters <matthew@centricular.com> | 2018-09-21 19:45:12 +1000 |
| commit | 07e9374eff7967713ab2dc47b1eb2843d0bfcaa0 (patch) | |
| tree | e745fcad33ad9bdbdfe1be64b94804ccb7e3697d /gst-libs | |
| parent | cf46d49b1e94587b15093dfdf12431c1ab00ba57 (diff) | |
| download | gstreamer-plugins-bad-07e9374eff7967713ab2dc47b1eb2843d0bfcaa0.tar.gz | |
webrtcbin: add support for data channels based on SCTP
Mostly follows the W3C specification
https://www.w3.org/TR/webrtc/#peer-to-peer-data-api
With contributions from:
Mathieu Duponchelle <mathieu@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=794351
Diffstat (limited to 'gst-libs')
| -rw-r--r-- | gst-libs/gst/webrtc/webrtc_fwd.h | 53 |
1 files changed, 53 insertions, 0 deletions
diff --git a/gst-libs/gst/webrtc/webrtc_fwd.h b/gst-libs/gst/webrtc/webrtc_fwd.h index be0a3c334..2e50125f5 100644 --- a/gst-libs/gst/webrtc/webrtc_fwd.h +++ b/gst-libs/gst/webrtc/webrtc_fwd.h @@ -264,4 +264,57 @@ typedef enum /*< underscore_name=gst_webrtc_fec_type >*/ GST_WEBRTC_FEC_TYPE_ULP_RED, } GstWebRTCFECType; +/** + * GstWebRTCSCTPTransportState: + * GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new + * GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting + * GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected + * GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed + * + * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate">http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate</ulink> + */ +typedef enum /*< underscore_name=gst_webrtc_sctp_transport_state >*/ +{ + GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW, + GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING, + GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED, + GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED, +} GstWebRTCSCTPTransportState; + +/** + * GstWebRTCPriorityType: + * GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low + * GST_WEBRTC_PRIORITY_TYPE_LOW: low + * GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium + * GST_WEBRTC_PRIORITY_TYPE_HIGH: high + * + * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype">http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype</ulink> + */ +typedef enum /*< underscore_name=gst_webrtc_priority_type >*/ +{ + GST_WEBRTC_PRIORITY_TYPE_VERY_LOW = 1, + GST_WEBRTC_PRIORITY_TYPE_LOW, + GST_WEBRTC_PRIORITY_TYPE_MEDIUM, + GST_WEBRTC_PRIORITY_TYPE_HIGH, +} GstWebRTCPriorityType; + +/** + * GstWebRTCDataChannelState: + * GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new + * GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection + * GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open + * GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing + * GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed + * + * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate">http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate</ulink> + */ +typedef enum /*< underscore_name=gst_webrtc_data_channel_state >*/ +{ + GST_WEBRTC_DATA_CHANNEL_STATE_NEW, + GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING, + GST_WEBRTC_DATA_CHANNEL_STATE_OPEN, + GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING, + GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED, +} GstWebRTCDataChannelState; + #endif /* __GST_WEBRTC_FWD_H__ */ |