| /third_party/libwebsockets/lib/secure-streams/ |
| D | README.md | 497 ### `http_method`
502 ### `http_expect`
512 ### `http_fail_redirect`
519 ### `http_url`
531 ### `http_resp_map`
546 ### `http_auth_header`
554 ### `http_dsn_header`
562 ### `http_fwv_header`
570 ### `http_devtype_header`
578 ### `http_auth_preamble`
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| /third_party/gstreamer/gstplugins_base/gst-libs/gst/rtsp/ |
| D | gstrtsptransport.c | 156 gst_rtsp_transport_new (GstRTSPTransport ** transport) in gst_rtsp_transport_new()
178 gst_rtsp_transport_init (GstRTSPTransport * transport) in gst_rtsp_transport_init()
249 gst_rtsp_transport_get_media_type (GstRTSPTransport * transport, in gst_rtsp_transport_get_media_type()
267 get_default_lower_trans (GstRTSPTransport * transport) in get_default_lower_trans()
316 parse_mode (GstRTSPTransport * transport, const gchar * str) in parse_mode()
388 rtsp_transport_mode_as_text (const GstRTSPTransport * transport) in rtsp_transport_mode_as_text()
400 rtsp_transport_profile_as_text (const GstRTSPTransport * transport) in rtsp_transport_profile_as_text()
412 rtsp_transport_ltrans_as_text (const GstRTSPTransport * transport) in rtsp_transport_ltrans_as_text()
443 gst_rtsp_transport_parse (const gchar * str, GstRTSPTransport * transport) in gst_rtsp_transport_parse()
599 gst_rtsp_transport_as_text (GstRTSPTransport * transport) in gst_rtsp_transport_as_text()
[all …]
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| D | gstrtspextension.c | 170 GstRTSPLowerTrans protocols, gchar ** transport) in gst_rtsp_extension_get_transports()
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| /third_party/spirv-tools/utils/vscode/ |
| D | extension.js | 32 run: { command: serverModule, transport: langClient.stdio }, property
33 debug: { command: serverModule, transport: langClient.stdio, options: debugOptions } property
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| /third_party/skia/third_party/externals/spirv-tools/utils/vscode/ |
| D | extension.js | 32 run: { command: serverModule, transport: langClient.stdio }, property
33 debug: { command: serverModule, transport: langClient.stdio, options: debugOptions } property
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| /third_party/skia/third_party/externals/swiftshader/third_party/SPIRV-Tools/utils/vscode/ |
| D | extension.js | 32 run: { command: serverModule, transport: langClient.stdio }, property
33 debug: { command: serverModule, transport: langClient.stdio, options: debugOptions } property
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| /third_party/gstreamer/gstplugins_bad/ext/webrtc/ |
| D | transportreceivebin.c | 152 GstWebRTCDTLSTransport *transport; in transport_receive_bin_set_receive_state() local
177 _on_notify_ice_connection_state (GstWebRTCICETransport * transport, in _on_notify_ice_connection_state()
313 GstWebRTCDTLSTransport *transport; in transport_receive_bin_constructed() local
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| D | gstwebrtcstats.c | 566 GstWebRTCICETransport * transport, const GstStructure * twcc_stats, in _get_stats_from_ice_transport()
642 GstWebRTCDTLSTransport * transport, const GstStructure * twcc_stats, in _get_stats_from_dtls_transport()
704 GstWebRTCDTLSTransport *transport; in _get_stats_from_transport_channel() local
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| D | webrtcsctptransport.h | 47 GstWebRTCDTLSTransport *transport; member
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| D | transportstream.h | 62 GstWebRTCDTLSTransport *transport; member
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| /third_party/python/Lib/asyncio/ |
| D | streams.py | 220 def connection_made(self, transport): argument
305 def __init__(self, transport, protocol, reader, loop): argument
322 def transport(self): member in StreamWriter
439 def set_transport(self, transport): argument
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| D | subprocess.py | 40 def connection_made(self, transport): argument
116 def __init__(self, transport, protocol, loop): argument
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| D | protocols.py | 21 def connection_made(self, transport): argument
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| /third_party/ffmpeg/libavformat/ |
| D | tls_gnutls.c | 108 static ssize_t gnutls_url_pull(gnutls_transport_ptr_t transport, in gnutls_url_pull()
124 static ssize_t gnutls_url_push(gnutls_transport_ptr_t transport, in gnutls_url_push()
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| /third_party/gstreamer/gstplugins_bad/sys/bluez/ |
| D | gstavdtputil.h | 56 gchar *transport; member
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| D | gsta2dpsink.h | 52 gchar *transport; member
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| /third_party/python/Lib/test/test_asyncio/ |
| D | test_sendfile.py | 43 def connection_made(self, transport): argument
78 def connection_made(self, transport): argument
160 def reduce_send_buffer_size(self, sock, transport=None): argument
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| D | test_events.py | 94 def connection_made(self, transport): argument
117 def connection_made(self, transport): argument
135 def connection_made(self, transport): argument
168 def connection_made(self, transport): argument
203 def connection_made(self, transport): argument
232 def connection_made(self, transport): argument
1142 def connection_made(self, transport): argument
1181 def connection_made(self, transport): argument
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| /third_party/gstreamer/gstplugins_bad/gst-libs/gst/webrtc/ |
| D | webrtc-priv.h | 109 GstWebRTCDTLSTransport *transport; member
140 GstWebRTCDTLSTransport *transport; member
203 GstWebRTCICETransport *transport; member
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| D | dtlstransport.c | 64 gst_webrtc_dtls_transport_set_transport (GstWebRTCDTLSTransport * transport, in gst_webrtc_dtls_transport_set_transport()
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| /third_party/gstreamer/gstplugins_good/ext/jack/ |
| D | gstjackaudiosrc.h | 74 guint transport; member
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| D | gstjackaudiosink.h | 57 guint transport; member
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| /third_party/openssl/crypto/rand/ |
| D | rand_egd.c | 73 char* transport) in hpns_socket()
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| /third_party/pulseaudio/src/modules/rtp/ |
| D | rtsp_client.c | 80 char *transport; member
554 int pa_rtsp_setup(pa_rtsp_client *c, const char *transport) { in pa_rtsp_setup()
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| /third_party/gstreamer/gstplugins_good/gst/rtsp/ |
| D | gstrtspext.c | 213 GstRTSPLowerTrans protocols, gchar ** transport) in gst_rtsp_ext_list_get_transports()
|