/third_party/gstreamer/gstplugins_bad/ext/webrtc/ |
D | gstwebrtcbin.c | 138 _have_nice_elements (GstWebRTCBin * webrtc) in _have_nice_elements() 164 _have_sctp_elements (GstWebRTCBin * webrtc) in _have_sctp_elements() 190 _have_dtls_elements (GstWebRTCBin * webrtc) in _have_dtls_elements() 282 _get_pending_sink_transceiver (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad) in _get_pending_sink_transceiver() 298 GstWebRTCBin *webrtc = GST_WEBRTC_BIN (parent); in gst_webrtcbin_sink_event() local 561 _find_ice_stream_for_session (GstWebRTCBin * webrtc, guint session_id) in _find_ice_stream_for_session() 582 _add_ice_stream_item (GstWebRTCBin * webrtc, guint session_id, in _add_ice_stream_item() 596 _find_transceiver (GstWebRTCBin * webrtc, gconstpointer data, in _find_transceiver() 628 _find_transceiver_for_mline (GstWebRTCBin * webrtc, guint mlineindex) in _find_transceiver_for_mline() 646 _find_transport (GstWebRTCBin * webrtc, gconstpointer data, in _find_transport() [all …]
|
D | utils.c | 52 _get_latest_offer (GstWebRTCBin * webrtc) in _get_latest_offer() 67 _get_latest_answer (GstWebRTCBin * webrtc) in _get_latest_answer() 82 _get_latest_sdp (GstWebRTCBin * webrtc) in _get_latest_sdp() 95 _get_latest_self_generated_sdp (GstWebRTCBin * webrtc) in _get_latest_self_generated_sdp()
|
D | gstwebrtcstats.c | 74 _get_peer_connection_stats (GstWebRTCBin * webrtc) in _get_peer_connection_stats() 108 _get_stats_from_remote_rtp_source_stats (GstWebRTCBin * webrtc, in _get_stats_from_remote_rtp_source_stats() 208 _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc, in _get_stats_from_rtp_source_stats() 565 _get_stats_from_ice_transport (GstWebRTCBin * webrtc, in _get_stats_from_ice_transport() 641 _get_stats_from_dtls_transport (GstWebRTCBin * webrtc, in _get_stats_from_dtls_transport() 700 _get_stats_from_transport_channel (GstWebRTCBin * webrtc, in _get_stats_from_transport_channel() 767 _get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, in _get_codec_stats_from_pad() 863 _get_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s) in _get_stats_from_pad() 893 gst_webrtc_bin_create_stats (GstWebRTCBin * webrtc, GstPad * pad) in gst_webrtc_bin_create_stats()
|
D | transportstream.c | 208 GstWebRTCBin *webrtc; in transport_stream_constructed() local 315 transport_stream_new (GstWebRTCBin * webrtc, guint session_id) in transport_stream_new()
|
D | webrtctransceiver.c | 213 webrtc_transceiver_new (GstWebRTCBin * webrtc, GstWebRTCRTPSender * sender, in webrtc_transceiver_new()
|
D | webrtcsctptransport.c | 67 _execute_task (GstWebRTCBin * webrtc, struct task *task) in _execute_task()
|
D | gstwebrtcbin.h | 153 GstWebRTCBin *webrtc; member
|
D | webrtcdatachannel.c | 225 _execute_task (GstWebRTCBin * webrtc, struct task *task) in _execute_task()
|
/third_party/gstreamer/gstplugins_bad/gst-libs/gst/webrtc/ |
D | rtptransceiver.c | 73 GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object); in gst_webrtc_rtp_transceiver_set_property() local 105 GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object); in gst_webrtc_rtp_transceiver_get_property() local 145 GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object); in gst_webrtc_rtp_transceiver_constructed() local 156 GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object); in gst_webrtc_rtp_transceiver_dispose() local 175 GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object); in gst_webrtc_rtp_transceiver_finalize() local 300 gst_webrtc_rtp_transceiver_init (GstWebRTCRTPTransceiver * webrtc) in gst_webrtc_rtp_transceiver_init()
|
D | dtlstransport.c | 86 GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object); in gst_webrtc_dtls_transport_set_property() local 109 GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object); in gst_webrtc_dtls_transport_get_property() local 140 GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object); in gst_webrtc_dtls_transport_finalize() local 154 GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (user_data); in on_connection_state_changed() local 183 GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object); in gst_webrtc_dtls_transport_constructed() local 257 gst_webrtc_dtls_transport_init (GstWebRTCDTLSTransport * webrtc) in gst_webrtc_dtls_transport_init()
|
D | icetransport.c | 107 GstWebRTCICETransport *webrtc = GST_WEBRTC_ICE_TRANSPORT (object); in gst_webrtc_ice_transport_set_property() local 123 GstWebRTCICETransport *webrtc = GST_WEBRTC_ICE_TRANSPORT (object); in gst_webrtc_ice_transport_get_property() local 208 gst_webrtc_ice_transport_init (GstWebRTCICETransport * webrtc) in gst_webrtc_ice_transport_init()
|
D | rtpreceiver.c | 89 GstWebRTCRTPReceiver *webrtc = GST_WEBRTC_RTP_RECEIVER (object); in gst_webrtc_rtp_receiver_finalize() local 123 gst_webrtc_rtp_receiver_init (GstWebRTCRTPReceiver * webrtc) in gst_webrtc_rtp_receiver_init()
|
D | rtpsender.c | 174 gst_webrtc_rtp_sender_init (GstWebRTCRTPSender * webrtc) in gst_webrtc_rtp_sender_init()
|
/third_party/gstreamer/gstplugins_bad/tests/examples/webrtc/ |
D | webrtctransceiver.c | 62 _webrtc_pad_added (GstElement * webrtc, GstPad * new_pad, GstElement * pipe) in _webrtc_pad_added() 139 _on_ice_candidate (GstElement * webrtc, guint mlineindex, gchar * candidate, in _on_ice_candidate() 146 _on_new_transceiver (GstElement * webrtc, GstWebRTCRTPTransceiver * trans) in _on_new_transceiver() 154 add_fec_to_offer (GstElement * webrtc) in add_fec_to_offer()
|
D | webrtcrenego.c | 19 GstElement *receive, *webrtc; in _element_message() local 109 _webrtc_pad_added (GstElement * webrtc, GstPad * new_pad, GstElement * pipe) in _webrtc_pad_added() 193 _on_ice_candidate (GstElement * webrtc, guint mlineindex, gchar * candidate, in _on_ice_candidate()
|
D | webrtc.c | 62 _webrtc_pad_added (GstElement * webrtc, GstPad * new_pad, GstElement * pipe) in _webrtc_pad_added() 139 _on_ice_candidate (GstElement * webrtc, guint mlineindex, gchar * candidate, in _on_ice_candidate()
|
D | webrtcbidirectional.c | 62 _webrtc_pad_added (GstElement * webrtc, GstPad * new_pad, GstElement * pipe) in _webrtc_pad_added() 146 _on_ice_candidate (GstElement * webrtc, guint mlineindex, gchar * candidate, in _on_ice_candidate()
|
D | webrtcswap.c | 62 _webrtc_pad_added (GstElement * webrtc, GstPad * new_pad, GstElement * pipe) in _webrtc_pad_added() 159 _on_ice_candidate (GstElement * webrtc, guint mlineindex, gchar * candidate, in _on_ice_candidate()
|
/third_party/gstreamer/gstplugins_bad/tests/examples/ |
D | meson.build | 16 subdir('webrtc') subdir
|
/third_party/gstreamer/gstplugins_bad/gst-libs/gst/ |
D | meson.build | 21 subdir('webrtc') subdir
|
/third_party/pulseaudio/src/modules/echo-cancel/ |
D | echo-cancel.h | 74 } webrtc; member
|
/third_party/gstreamer/gstplugins_bad/ext/ |
D | meson.build | 67 subdir('webrtc') subdir
|
/third_party/gstreamer/gstplugins_bad/ |
D | meson_options.txt | 168 option('webrtc', type : 'feature', value : 'auto', description : 'WebRTC audio/video network bin pl… feature
|
/third_party/gstreamer/gstplugins_bad/tests/check/elements/ |
D | webrtcbin.c | 385 _on_negotiation_needed (GstElement * webrtc, struct test_webrtc *t) in _on_negotiation_needed() 397 _on_ice_candidate (GstElement * webrtc, guint mlineindex, gchar * candidate, in _on_ice_candidate() 414 _on_pad_added (GstElement * webrtc, GstPad * new_pad, struct test_webrtc *t) in _on_pad_added() 423 _on_data_channel (GstElement * webrtc, GObject * data_channel, in _on_data_channel()
|
/third_party/gstreamer/gstplugins_bad/docs/plugins/ |
D | gst_plugins_cache.json | 229455 "webrtc": { object
|