webrtcbin: an element that handles the transport aspects of webrtc connections

SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/

The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer.  In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.

The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.

With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792523

4 files changed, 132 insertions, 0 deletions

diff --git a/docs/libs/Makefile.am b/docs/libs/Makefile.am
index dfc20ff53..ef2d37f8e 100644
--- a/docs/libs/Makefile.am
+++ b/docs/libs/Makefile.am
@@ -61,6 +61,7 @@ GTKDOC_LIBS = \
 	$(top_builddir)/gst-libs/gst/insertbin/libgstinsertbin-@GST_API_VERSION@.la \
 	$(top_builddir)/gst-libs/gst/mpegts/libgstmpegts-@GST_API_VERSION@.la \
 	$(top_builddir)/gst-libs/gst/player/libgstplayer-@GST_API_VERSION@.la \
+	$(top_builddir)/gst-libs/gst/webrtc/libgstwebrtc-@GST_API_VERSION@.la \
   $(GST_BASE_LIBS)
 
 # If you need to override some of the declarations, place them in this file
diff --git a/docs/libs/gst-plugins-bad-libs-docs.sgml b/docs/libs/gst-plugins-bad-libs-docs.sgml
index 530af2b4d..6a237c8da 100644
--- a/docs/libs/gst-plugins-bad-libs-docs.sgml
+++ b/docs/libs/gst-plugins-bad-libs-docs.sgml
@@ -73,6 +73,16 @@
       <xi:include href="xml/gstplayer-visualization.xml"/>
     </chapter>
 
+    <chapter id="webrtc">
+      <title>WebRTC Library</title>
+      <xi:include href="xml/gstwebrtc-dtlstransport.xml"/>
+      <xi:include href="xml/gstwebrtc-icetransport.xml"/>
+      <xi:include href="xml/gstwebrtc-receiver.xml"/>
+      <xi:include href="xml/gstwebrtc-sender.xml"/>
+      <xi:include href="xml/gstwebrtc-sessiondescription.xml"/>
+      <xi:include href="xml/gstwebrtc-transceiver.xml"/>
+    </chapter>
+
     <chapter>
       <title>Interfaces</title>
       <xi:include href="xml/gstphotography.xml" />
diff --git a/docs/libs/gst-plugins-bad-libs-sections.txt b/docs/libs/gst-plugins-bad-libs-sections.txt
index 7becdeb68..7231614cc 100644
--- a/docs/libs/gst-plugins-bad-libs-sections.txt
+++ b/docs/libs/gst-plugins-bad-libs-sections.txt
@@ -1065,3 +1065,104 @@ GstPlayerSubtitleInfoClass
 gst_player_subtitle_info_get_type
 </SECTION>
 
+
+<SECTION>
+<FILE>gstwebrtc-dtlstransport</FILE>
+GstWebRTCDTLSTransportState
+
+gst_webrtc_dtls_transport_new
+
+<SUBSECTION Standard>
+GST_TYPE_WEBRTC_DTLS_TRANSPORT
+gst_webrtc_dtls_transport_get_type
+GstWebRTCDTLSTransport
+GST_WEBRTC_DTLS_TRANSPORT
+GST_IS_WEBRTC_DTLS_TRANSPORT
+GstWebRTCDTLSTransportClass
+GST_WEBRTC_DTLS_TRANSPORT_CLASS
+GST_WEBRTC_DTLS_TRANSPORT_GET_CLASS
+GST_IS_WEBRTC_DTLS_TRANSPORT_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>gstwebrtc-icetransport</FILE>
+GstWebRTCIceRole
+GstWebRTCICEConnectionState
+GstWebRTCICEGatheringState
+
+
+
+<SUBSECTION Standard>
+GST_TYPE_WEBRTC_ICE_TRANSPORT
+gst_webrtc_ice_transport_get_type
+GstWebRTCICETransport
+GST_WEBRTC_ICE_TRANSPORT
+GST_IS_WEBRTC_ICE_TRANSPORT
+GstWebRTCICETransportClass
+GST_WEBRTC_ICE_TRANSPORT_CLASS
+GST_WEBRTC_ICE_TRANSPORT_GET_CLASS
+GST_IS_WEBRTC_ICE_TRANSPORT_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>gstwebrtc-receiver</FILE>
+gst_webrtc_rtp_receiver_new
+gst_webrtc_rtp_receiver_get_parameters
+gst_webrtc_rtp_receiver_set_parameters
+gst_webrtc_rtp_receiver_set_rtcp_transport
+gst_webrtc_rtp_receiver_set_transport
+<SUBSECTION Standard>
+GST_TYPE_WEBRTC_RTP_RECEIVER
+gst_webrtc_rtp_receiver_get_type
+GstWebRTCRTPReceiver
+GST_WEBRTC_RTP_RECEIVER
+GST_IS_WEBRTC_RTP_RECEIVER
+GstWebRTCRTPReceiverClass
+GST_WEBRTC_RTP_RECEIVER_CLASS
+GST_WEBRTC_RTP_RECEIVER_GET_CLASS
+GST_IS_WEBRTC_RTP_RECEIVER_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>gstwebrtc-sender</FILE>
+gst_webrtc_rtp_sender_new
+gst_webrtc_rtp_sender_get_parameters
+gst_webrtc_rtp_sender_set_parameters
+gst_webrtc_rtp_sender_set_rtcp_transport
+gst_webrtc_rtp_sender_set_transport
+<SUBSECTION Standard>
+GST_TYPE_WEBRTC_RTP_SENDER
+gst_webrtc_rtp_sender_get_type
+GstWebRTCRTPSender
+GST_WEBRTC_RTP_SENDER
+GST_IS_WEBRTC_RTP_SENDER
+GstWebRTCRTPSenderClass
+GST_WEBRTC_RTP_SENDER_CLASS
+GST_WEBRTC_RTP_SENDER_GET_CLASS
+GST_IS_WEBRTC_RTP_SENDER_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>gstwebrtc-sessiondescription</FILE>
+GstWebRTCSessionDescription
+gst_webrtc_session_description_new
+gst_webrtc_session_description_copy
+gst_webrtc_session_description_free
+<SUBSECTION Standard>
+gst_webrtc_session_description_get_type
+GST_TYPE_WEBRTC_SESSION_DESCRIPTION
+</SECTION>
+
+<SECTION>
+<FILE>gstwebrtc-transceiver</FILE>
+<SUBSECTION Standard>
+GST_TYPE_WEBRTC_RTP_TRANSCEIVER
+gst_webrtc_rtp_transceiver_get_type
+GstWebRTCRTPTransceiver
+GST_WEBRTC_RTP_TRANSCEIVER
+GST_IS_WEBRTC_RTP_TRANSCEIVER
+GstWebRTCRTPTransceiverClass
+GST_WEBRTC_RTP_TRANSCEIVER_CLASS
+GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS
+GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS
+</SECTION>
diff --git a/docs/libs/gst-plugins-bad-libs.types b/docs/libs/gst-plugins-bad-libs.types
index 1b3a55534..7b4851c42 100644
--- a/docs/libs/gst-plugins-bad-libs.types
+++ b/docs/libs/gst-plugins-bad-libs.types
@@ -7,6 +7,7 @@
 #include <gst/insertbin/gstinsertbin.h>
 #include <gst/mpegts/mpegts.h>
 #include <gst/player/player.h>
+#include <gst/webrtc/webrtc.h>
 
 gst_audio_aggregator_get_type
 gst_audio_aggregator_pad_get_type
@@ -49,3 +50,22 @@ gst_player_video_overlay_video_renderer_get_type
 gst_player_video_renderer_get_type
 gst_player_visualization_get_type
 
+
+gst_webrtc_dtls_setup_get_type
+gst_webrtc_dtls_transport_get_type
+gst_webrtc_dtls_transport_state_get_type
+gst_webrtc_ice_component_get_type
+gst_webrtc_ice_connection_state_get_type
+gst_webrtc_ice_gathering_state_get_type
+gst_webrtc_ice_role_get_type
+gst_webrtc_sdp_type_get_type
+gst_webrtc_ice_transport_get_type
+gst_webrtc_peer_connection_state_get_type
+gst_webrtc_rtp_receiver_get_type
+gst_webrtc_rtp_sender_get_type
+gst_webrtc_session_description_get_type
+gst_webrtc_signaling_state_get_type
+gst_webrtc_rtp_transceiver_direction_get_type
+gst_webrtc_rtp_transceiver_get_type
+gst_webrtc_stats_type_get_type
+