gstreamer/gst-libs/gst/webrtc/webrtc_fwd.h

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/* 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>
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/**
* SECTION:webrtc_fwd.h
* @title: GstWebRTC Enumerations
*/
#ifndef GST_WEBRTC_API
libs: fix API export/import and 'inconsistent linkage' on MSVC For each lib we build export its own API in headers when we're building it, otherwise import the API from the headers. This fixes linker warnings on Windows when building with MSVC. The problem was that we had defined all GST_*_API decorators unconditionally to GST_EXPORT. This was intentional and only supposed to be temporary, but caused linker warnings because we tell the linker that we want to export all symbols even those from externall DLLs, and when the linker notices that they were in external DLLS and not present locally it warns. What we need to do when building each library is: export the library's own symbols and import all other symbols. To this end we define e.g. BUILDING_GST_FOO and then we define the GST_FOO_API decorator either to export or to import symbols depending on whether BUILDING_GST_FOO is set or not. That way external users of each library API automatically get the import. While we're at it, add new GST_API_EXPORT in config.h and use that for GST_*_API decorators instead of GST_EXPORT. The right export define depends on the toolchain and whether we're using -fvisibility=hidden or not, so it's better to set it to the right thing directly than hard-coding a compiler whitelist in the public header. We put the export define into config.h instead of passing it via the command line to the compiler because it might contain spaces and brackets and in the autotools scenario we'd have to pass that through multiple layers of plumbing and Makefile/shell escaping and we're just not going to be *that* lucky. The export define is only used if we're compiling our lib, not by external users of the lib headers, so it's not a problem to put it into config.h Also, this means all .c files of libs need to include config.h to get the export marker defined, so fix up a few that didn't include config.h. This commit depends on a common submodule commit that makes gst-glib-gen.mak add an #include "config.h" to generated enum/marshal .c files for the autotools build. https://bugzilla.gnome.org/show_bug.cgi?id=797185
2018-09-24 10:52:22 +00:00
# ifdef BUILDING_GST_WEBRTC
# define GST_WEBRTC_API GST_API_EXPORT /* from config.h */
# else
# define GST_WEBRTC_API GST_API_IMPORT
# endif
#endif
#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;
typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel;
typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass;
/**
* GstWebRTCDTLSTransportState:
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* @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:
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* @GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
* @GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
* @GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
*
* See <http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate>
*/
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:
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* @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 <http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate>
*/
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:
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* @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 <http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate>
*/
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:
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* @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 <http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate>
*/
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:
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* @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:
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* @GST_WEBRTC_ICE_COMPONENT_RTP: RTP component
* @GST_WEBRTC_ICE_COMPONENT_RTCP: RTCP component
*/
typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
{
GST_WEBRTC_ICE_COMPONENT_RTP,
GST_WEBRTC_ICE_COMPONENT_RTCP,
} GstWebRTCICEComponent;
/**
* GstWebRTCSDPType:
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* @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 <http://w3c.github.io/webrtc-pc/#rtcsdptype>
*/
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;
/**
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* 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:
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* @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:
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* @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;
/**
* GstWebRTCFECType:
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* @GST_WEBRTC_FEC_TYPE_NONE: none
* @GST_WEBRTC_FEC_TYPE_ULP_RED: ulpfec + red
*
* Since: 1.14.1
*/
typedef enum /*< underscore_name=gst_webrtc_fec_type >*/
{
GST_WEBRTC_FEC_TYPE_NONE,
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 <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate>
*
* Since: 1.16
*/
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 <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
*
* Since: 1.16
*/
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 <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate>
*
* Since: 1.16
*/
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;
/**
* GstWebRTCBundlePolicy:
* GST_WEBRTC_BUNDLE_POLICY_NONE: none
* GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced
* GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat
* GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
*
* See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
* for more information.
*
* Since: 1.16
*/
typedef enum /*<underscore_name=gst_webrtc_bundle_policy>*/
{
GST_WEBRTC_BUNDLE_POLICY_NONE,
GST_WEBRTC_BUNDLE_POLICY_BALANCED,
GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT,
GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE,
} GstWebRTCBundlePolicy;
/**
* GstWebRTCICETransportPolicy:
* GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all
* GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
*
* See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
* for more information.
*
* Since: 1.16
*/
typedef enum /*<underscore_name=gst_webrtc_ice_transport_policy>*/
{
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL,
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY,
} GstWebRTCICETransportPolicy;
/**
* GstWebRTCKind:
* @GST_WEBRTC_KIND_UNKNOWN: Kind has not yet been set
* @GST_WEBRTC_KIND_AUDIO: Kind is audio
* @GST_WEBRTC_KIND_VIDEO: Kind is audio
*
* https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind
*
* Since: 1.20
*/
typedef enum /*<underscore_name=gst_webrtc_kind>*/
{
GST_WEBRTC_KIND_UNKNOWN,
GST_WEBRTC_KIND_AUDIO,
GST_WEBRTC_KIND_VIDEO,
} GstWebRTCKind;
#endif /* __GST_WEBRTC_FWD_H__ */