1 files changed, 251 insertions, 0 deletions
diff --git a/gst-libs/gst/webrtc/webrtc_fwd.h b/gst-libs/gst/webrtc/webrtc_fwd.h
new file mode 100644
index 000000000..48c9bdab1
--- /dev/null
+++ b/gst-libs/gst/webrtc/webrtc_fwd.h
@@ -0,0 +1,251 @@
+/* GStreamer
+ * Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_WEBRTC_FWD_H__
+#define __GST_WEBRTC_FWD_H__
+
+#ifndef GST_USE_UNSTABLE_API
+#warning "The WebRTC library from gst-plugins-bad is unstable API and may change in future."
+#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
+#endif
+
+#include <gst/gst.h>
+#include <gst/webrtc/webrtc-enumtypes.h>
+
+typedef struct _GstWebRTCDTLSTransport GstWebRTCDTLSTransport;
+typedef struct _GstWebRTCDTLSTransportClass GstWebRTCDTLSTransportClass;
+
+typedef struct _GstWebRTCICETransport GstWebRTCICETransport;
+typedef struct _GstWebRTCICETransportClass GstWebRTCICETransportClass;
+
+typedef struct _GstWebRTCRTPReceiver GstWebRTCRTPReceiver;
+typedef struct _GstWebRTCRTPReceiverClass GstWebRTCRTPReceiverClass;
+
+typedef struct _GstWebRTCRTPSender GstWebRTCRTPSender;
+typedef struct _GstWebRTCRTPSenderClass GstWebRTCRTPSenderClass;
+
+typedef struct _GstWebRTCSessionDescription GstWebRTCSessionDescription;
+
+typedef struct _GstWebRTCRTPTransceiver GstWebRTCRTPTransceiver;
+typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass;
+
+/**
+ * GstWebRTCDTLSTransportState:
+ * GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
+ * GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
+ * GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
+ * GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
+ * GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected
+ */
+typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/
+{
+  GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW,
+  GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED,
+  GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED,
+  GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING,
+  GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED,
+} GstWebRTCDTLSTransportState;
+
+/**
+ * GstWebRTCICEGatheringState:
+ * GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
+ * GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
+ * GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
+ *
+ * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink>
+ */
+typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/
+{
+  GST_WEBRTC_ICE_GATHERING_STATE_NEW,
+  GST_WEBRTC_ICE_GATHERING_STATE_GATHERING,
+  GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE,
+} GstWebRTCICEGatheringState; /*< underscore_name=gst_webrtc_ice_gathering_state >*/
+
+/**
+ * GstWebRTCICEConnectionState:
+ * GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
+ * GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
+ * GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
+ * GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
+ * GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
+ * GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
+ * GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
+ *
+ * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink>
+ */
+typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/
+{
+  GST_WEBRTC_ICE_CONNECTION_STATE_NEW,
+  GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING,
+  GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED,
+  GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED,
+  GST_WEBRTC_ICE_CONNECTION_STATE_FAILED,
+  GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED,
+  GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED,
+} GstWebRTCICEConnectionState;
+
+/**
+ * GstWebRTCSignalingState:
+ * GST_WEBRTC_SIGNALING_STATE_STABLE: stable
+ * GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
+ * GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
+ * GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
+ * GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
+ * GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
+ *
+ * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate">http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate</ulink>
+ */
+typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/
+{
+  GST_WEBRTC_SIGNALING_STATE_STABLE,
+  GST_WEBRTC_SIGNALING_STATE_CLOSED,
+  GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER,
+  GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER,
+  GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER,
+  GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER,
+} GstWebRTCSignalingState;
+
+/**
+ * GstWebRTCPeerConnectionState:
+ * GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
+ * GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
+ * GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
+ * GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
+ * GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
+ * GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
+ *
+ * See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink>
+ */
+typedef enum /*< underscore_name=gst_webrtc_peer_connection_state >*/
+{
+  GST_WEBRTC_PEER_CONNECTION_STATE_NEW,
+  GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING,
+  GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED,
+  GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED,
+  GST_WEBRTC_PEER_CONNECTION_STATE_FAILED,
+  GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED,
+} GstWebRTCPeerConnectionState;
+
+/**
+ * GstWebRTCIceRole:
+ * GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
+ * GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
+ */
+typedef enum /*< underscore_name=gst_webrtc_ice_role >*/
+{
+  GST_WEBRTC_ICE_ROLE_CONTROLLED,
+  GST_WEBRTC_ICE_ROLE_CONTROLLING,
+} GstWebRTCIceRole;
+
+/**
+ * GstWebRTCIceComponent:
+ * GST_WEBRTC_ICE_COMPONENT_RTP,
+ * GST_WEBRTC_ICE_COMPONENT_RTCP,
+ */
+typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
+{
+  GST_WEBRTC_ICE_COMPONENT_RTP,
+  GST_WEBRTC_ICE_COMPONENT_RTCP,
+} GstWebRTCICEComponent;
+
+/**
+ * GstWebRTCSDPType:
+ * GST_WEBRTC_SDP_TYPE_OFFER: offer
+ * GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
+ * GST_WEBRTC_SDP_TYPE_ANSWER: answer
+ * GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
+ *
+ * See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink>
+ */
+typedef enum /*< underscore_name=gst_webrtc_sdp_type >*/
+{
+  GST_WEBRTC_SDP_TYPE_OFFER = 1,
+  GST_WEBRTC_SDP_TYPE_PRANSWER,
+  GST_WEBRTC_SDP_TYPE_ANSWER,
+  GST_WEBRTC_SDP_TYPE_ROLLBACK,
+} GstWebRTCSDPType;
+
+/**
+ * GstWebRTCRtpTransceiverDirection:
+ * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: none
+ * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: inactive
+ * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: sendonly
+ * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: recvonly
+ * GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: sendrecv
+ */
+typedef enum /*< underscore_name=gst_webrtc_rtp_transceiver_direction >*/
+{
+  GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE,
+  GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE,
+  GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY,
+  GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY,
+  GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV,
+} GstWebRTCRTPTransceiverDirection;
+
+/**
+ * GstWebRTCDTLSSetup:
+ * GST_WEBRTC_DTLS_SETUP_NONE: none
+ * GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
+ * GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
+ * GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
+ */
+typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/
+{
+  GST_WEBRTC_DTLS_SETUP_NONE,
+  GST_WEBRTC_DTLS_SETUP_ACTPASS,
+  GST_WEBRTC_DTLS_SETUP_ACTIVE,
+  GST_WEBRTC_DTLS_SETUP_PASSIVE,
+} GstWebRTCDTLSSetup;
+
+/**
+ * GstWebRTCStatsType:
+ * GST_WEBRTC_STATS_CODEC: codec
+ * GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
+ * GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
+ * GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
+ * GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
+ * GST_WEBRTC_STATS_CSRC: csrc
+ * GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
+ * GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
+ * GST_WEBRTC_STATS_STREAM: stream
+ * GST_WEBRTC_STATS_TRANSPORT: transport
+ * GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
+ * GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
+ * GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
+ * GST_WEBRTC_STATS_CERTIFICATE: certificate
+ */
+typedef enum /*< underscore_name=gst_webrtc_stats_type >*/
+{
+  GST_WEBRTC_STATS_CODEC = 1,
+  GST_WEBRTC_STATS_INBOUND_RTP,
+  GST_WEBRTC_STATS_OUTBOUND_RTP,
+  GST_WEBRTC_STATS_REMOTE_INBOUND_RTP,
+  GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP,
+  GST_WEBRTC_STATS_CSRC,
+  GST_WEBRTC_STATS_PEER_CONNECTION,
+  GST_WEBRTC_STATS_DATA_CHANNEL,
+  GST_WEBRTC_STATS_STREAM,
+  GST_WEBRTC_STATS_TRANSPORT,
+  GST_WEBRTC_STATS_CANDIDATE_PAIR,
+  GST_WEBRTC_STATS_LOCAL_CANDIDATE,
+  GST_WEBRTC_STATS_REMOTE_CANDIDATE,
+  GST_WEBRTC_STATS_CERTIFICATE,
+} GstWebRTCStatsType;
+
+#endif /* __GST_WEBRTC_FWD_H__ */