[vala/staging] gstreamer-1.0: Add gstreamer-webrtc-1.0 bindings

[Rico Tzschichholz <ricotz src gnome org>]

commit e6d43a8d76ab11847669bba67161b22a001aacb7
Author: Rico Tzschichholz <ricotz ubuntu com>
Date:   Sun Mar 18 20:50:39 2018 +0100

    gstreamer-1.0: Add gstreamer-webrtc-1.0 bindings

 vapi/Makefile.am               |    6 +
 vapi/gstreamer-webrtc-1.0.deps |    2 +
 vapi/gstreamer-webrtc-1.0.vapi |  199 ++++++++++++++++++++++++++++++++++++++++
 3 files changed, 207 insertions(+), 0 deletions(-)
---
diff --git a/vapi/Makefile.am b/vapi/Makefile.am
index 5f112b2..db994e1 100644
--- a/vapi/Makefile.am
+++ b/vapi/Makefile.am
@@ -159,6 +159,8 @@ dist_vapi_DATA = \
        gstreamer-tag-1.0.deps \
        gstreamer-video-1.0.vapi \
        gstreamer-video-1.0.deps \
+       gstreamer-webrtc-1.0.vapi \
+       gstreamer-webrtc-1.0.deps \
        gtk+-2.0.deps \
        gtk+-2.0.vapi \
        gtk+-3.0.deps \
@@ -405,6 +407,7 @@ GSTREAMER_1_0_BINDINGS = \
        gstreamer-rtsp-server-1.0 \
        gstreamer-tag-1.0 \
        gstreamer-video-1.0 \
+       gstreamer-webrtc-1.0 \
        $(NULL)
 
 GIR_BINDINGS = \
@@ -679,6 +682,9 @@ gstreamer-tag-1.0:
 gstreamer-video-1.0:
        $(GENVAPI) --library $(srcdir)/gstreamer-video-1.0 --pkg gstreamer-base-1.0 --metadatadir 
$(METADATADIR) $(METADATADIR)/GstVideo-1.0-custom.vala $(GIRDIR)/GstVideo-1.0.gir
 
+gstreamer-webrtc-1.0:
+       $(GENVAPI) --library $(srcdir)/gstreamer-webrtc-1.0 --pkg gstreamer-base-1.0 --pkg gstreamer-sdp-1.0 
--metadatadir $(METADATADIR) $(GIRDIR)/GstWebRTC-1.0.gir
+
 gtk+-2.0:
        $(GENVAPI) --library $(srcdir)/gtk+-2.0 $(PACKAGESDIR)/gtk+-2.0/gtk+-2.0-custom.vala 
$(PACKAGESDIR)/gtk+-2.0/gtk+-2.0.gi
 
diff --git a/vapi/gstreamer-webrtc-1.0.deps b/vapi/gstreamer-webrtc-1.0.deps
new file mode 100644
index 0000000..1a6a33c
--- /dev/null
+++ b/vapi/gstreamer-webrtc-1.0.deps
@@ -0,0 +1,2 @@
+gstreamer-base-1.0
+gstreamer-sdp-1.0
diff --git a/vapi/gstreamer-webrtc-1.0.vapi b/vapi/gstreamer-webrtc-1.0.vapi
new file mode 100644
index 0000000..f8471e2
--- /dev/null
+++ b/vapi/gstreamer-webrtc-1.0.vapi
@@ -0,0 +1,199 @@
+/* gstreamer-webrtc-1.0.vapi generated by vapigen, do not modify. */
+
+[CCode (cprefix = "Gst", gir_namespace = "GstWebRTC", gir_version = "1.0", lower_case_cprefix = "gst_")]
+namespace Gst {
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_dtls_transport", 
type_id = "gst_webrtc_dtls_transport_get_type ()")]
+       public class WebRTCDTLSTransport : Gst.Object {
+               [CCode (array_length = false)]
+               public weak void* _padding[4];
+               public weak Gst.Element dtlssrtpdec;
+               public weak Gst.Element dtlssrtpenc;
+               public bool is_rtcp;
+               [CCode (has_construct_function = false)]
+               public WebRTCDTLSTransport (uint session_id, bool rtcp);
+               public void set_transport (Gst.WebRTCICETransport ice);
+               [NoAccessorMethod]
+               public string certificate { owned get; set; }
+               [NoAccessorMethod]
+               public bool client { get; set; }
+               [NoAccessorMethod]
+               public string remote_certificate { owned get; }
+               [NoAccessorMethod]
+               public bool rtcp { get; construct; }
+               [NoAccessorMethod]
+               public uint session_id { get; construct; }
+               [NoAccessorMethod]
+               public Gst.WebRTCDTLSTransportState state { get; }
+               [NoAccessorMethod]
+               public Gst.WebRTCICETransport transport { owned get; }
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_ice_transport", 
type_id = "gst_webrtc_ice_transport_get_type ()")]
+       public abstract class WebRTCICETransport : Gst.Object {
+               [CCode (array_length = false)]
+               public weak void* _padding[4];
+               public Gst.WebRTCICERole role;
+               public weak Gst.Element sink;
+               public weak Gst.Element src;
+               [CCode (has_construct_function = false)]
+               protected WebRTCICETransport ();
+               public void connection_state_change (Gst.WebRTCICEConnectionState new_state);
+               [NoWrapper]
+               public virtual bool gather_candidates ();
+               public void gathering_state_change (Gst.WebRTCICEGatheringState new_state);
+               public void new_candidate (uint stream_id, Gst.WebRTCICEComponent component, string attr);
+               public void selected_pair_change ();
+               [NoAccessorMethod]
+               public Gst.WebRTCICEComponent component { get; construct; }
+               [NoAccessorMethod]
+               public Gst.WebRTCICEGatheringState gathering_state { get; }
+               [NoAccessorMethod]
+               public Gst.WebRTCICEConnectionState state { get; }
+               public signal void on_new_candidate (string object);
+               public signal void on_selected_candidate_pair_change ();
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_receiver", type_id 
= "gst_webrtc_rtp_receiver_get_type ()")]
+       public class WebRTCRTPReceiver : Gst.Object {
+               [CCode (array_length = false)]
+               public weak void* _padding[4];
+               public weak Gst.WebRTCDTLSTransport rtcp_transport;
+               public weak Gst.WebRTCDTLSTransport transport;
+               [CCode (has_construct_function = false)]
+               public WebRTCRTPReceiver ();
+               public void set_rtcp_transport (Gst.WebRTCDTLSTransport transport);
+               public void set_transport (Gst.WebRTCDTLSTransport transport);
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_sender", type_id = 
"gst_webrtc_rtp_sender_get_type ()")]
+       public class WebRTCRTPSender : Gst.Object {
+               [CCode (array_length = false)]
+               public weak void* _padding[4];
+               public weak Gst.WebRTCDTLSTransport rtcp_transport;
+               public weak GLib.Array<void*> send_encodings;
+               public weak Gst.WebRTCDTLSTransport transport;
+               [CCode (has_construct_function = false)]
+               public WebRTCRTPSender ();
+               public void set_rtcp_transport (Gst.WebRTCDTLSTransport transport);
+               public void set_transport (Gst.WebRTCDTLSTransport transport);
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_transceiver", 
type_id = "gst_webrtc_rtp_transceiver_get_type ()")]
+       public abstract class WebRTCRTPTransceiver : Gst.Object {
+               [CCode (array_length = false)]
+               public weak void* _padding[4];
+               public weak Gst.Caps codec_preferences;
+               public Gst.WebRTCRTPTransceiverDirection current_direction;
+               public Gst.WebRTCRTPTransceiverDirection direction;
+               public weak string mid;
+               public uint mline;
+               public bool stopped;
+               [CCode (has_construct_function = false)]
+               protected WebRTCRTPTransceiver ();
+               [NoAccessorMethod]
+               public uint mlineindex { get; construct; }
+               [NoAccessorMethod]
+               public Gst.WebRTCRTPReceiver receiver { owned get; construct; }
+               [NoAccessorMethod]
+               public Gst.WebRTCRTPSender sender { owned get; construct; }
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", copy_function = "g_boxed_copy", free_function = 
"g_boxed_free", lower_case_csuffix = "webrtc_session_description", type_id = 
"gst_webrtc_session_description_get_type ()")]
+       [Compact]
+       public class WebRTCSessionDescription {
+               public weak Gst.SDP.Message sdp;
+               public Gst.WebRTCSDPType type;
+               [CCode (has_construct_function = false)]
+               public WebRTCSessionDescription (Gst.WebRTCSDPType type, Gst.SDP.Message sdp);
+               public Gst.WebRTCSessionDescription copy ();
+               public void free ();
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_DTLS_SETUP_", type_id = 
"gst_webrtc_dtls_setup_get_type ()")]
+       public enum WebRTCDTLSSetup {
+               NONE,
+               ACTPASS,
+               ACTIVE,
+               PASSIVE
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_DTLS_TRANSPORT_STATE_", 
type_id = "gst_webrtc_dtls_transport_state_get_type ()")]
+       public enum WebRTCDTLSTransportState {
+               NEW,
+               CLOSED,
+               FAILED,
+               CONNECTING,
+               CONNECTED
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_COMPONENT_", type_id = 
"gst_webrtc_ice_component_get_type ()")]
+       public enum WebRTCICEComponent {
+               RTP,
+               RTCP
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_CONNECTION_STATE_", 
type_id = "gst_webrtc_ice_connection_state_get_type ()")]
+       public enum WebRTCICEConnectionState {
+               NEW,
+               CHECKING,
+               CONNECTED,
+               COMPLETED,
+               FAILED,
+               DISCONNECTED,
+               CLOSED
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_GATHERING_STATE_", 
type_id = "gst_webrtc_ice_gathering_state_get_type ()")]
+       public enum WebRTCICEGatheringState {
+               NEW,
+               GATHERING,
+               COMPLETE
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_ROLE_", type_id = 
"gst_webrtc_ice_role_get_type ()")]
+       public enum WebRTCICERole {
+               CONTROLLED,
+               CONTROLLING
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_PEER_CONNECTION_STATE_", 
type_id = "gst_webrtc_peer_connection_state_get_type ()")]
+       public enum WebRTCPeerConnectionState {
+               NEW,
+               CONNECTING,
+               CONNECTED,
+               DISCONNECTED,
+               FAILED,
+               CLOSED
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_", 
type_id = "gst_webrtc_rtp_transceiver_direction_get_type ()")]
+       public enum WebRTCRTPTransceiverDirection {
+               NONE,
+               INACTIVE,
+               SENDONLY,
+               RECVONLY,
+               SENDRECV
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SDP_TYPE_", type_id = 
"gst_webrtc_sdp_type_get_type ()")]
+       public enum WebRTCSDPType {
+               OFFER,
+               PRANSWER,
+               ANSWER,
+               ROLLBACK
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SIGNALING_STATE_", type_id = 
"gst_webrtc_signaling_state_get_type ()")]
+       public enum WebRTCSignalingState {
+               STABLE,
+               CLOSED,
+               HAVE_LOCAL_OFFER,
+               HAVE_REMOTE_OFFER,
+               HAVE_LOCAL_PRANSWER,
+               HAVE_REMOTE_PRANSWER
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_STATS_", type_id = 
"gst_webrtc_stats_type_get_type ()")]
+       public enum WebRTCStatsType {
+               CODEC,
+               INBOUND_RTP,
+               OUTBOUND_RTP,
+               REMOTE_INBOUND_RTP,
+               REMOTE_OUTBOUND_RTP,
+               CSRC,
+               PEER_CONNECTION,
+               DATA_CHANNEL,
+               STREAM,
+               TRANSPORT,
+               CANDIDATE_PAIR,
+               LOCAL_CANDIDATE,
+               REMOTE_CANDIDATE,
+               CERTIFICATE
+       }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h")]
+       public static unowned string webrtc_sdp_type_to_string (Gst.WebRTCSDPType type);
+}