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gstreamer/gst/rtsp/URLS

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gst/rtsp/URLS: Added some test URLS. Original commit message from CVS: * gst/rtsp/URLS: Added some test URLS. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_loop), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: When creating streams, give access to the complete SDP. Fix some leaks. Collect and merge global stream properties in stream caps. Preliminary support for WMServer. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspconnection.h: Make connection interruptable. Refactor to make it reconnectable. Don't fail on short reads when reading data packets. * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port), (rtsp_url_get_port): * gst/rtsp/rtspurl.h: Add methods for getting/setting the port. * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_message_get_attribute_val), (sdp_media_get_attribute), (sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val), (sdp_media_get_format), (sdp_parse_line), (sdp_message_parse_buffer): Fix headers. Add methods for getting multiple attributes with the same name. Increase buffer size when parsing. Fix parsing of a=foo fields. * gst/rtsp/test.c: (main): Update to new connection API. * gst/rtsp/rtspmessage.c: (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/rtsptransport.c: (rtsp_transport_free): * gst/rtsp/rtsptransport.h: * gst/rtsp/sdp.h: * gst/rtsp/sdpmessage.h: * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtsp.h: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtpdec.h: * gst/rtsp/rtsp.h: * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: Dual licensed under MIT and LGPL now.
2006-09-20 16:06:27 +00:00
Some test URLS:
SVQ3 video:
rtsp://cumulus.creative.auckland.ac.nz/~shado/nelson_iv_512k.mov
gst/rtsp/URLS: Add some more URLs. Original commit message from CVS: * gst/rtsp/URLS: Add some more URLs. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_loop), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Add timeout property to control UDP timeouts. Fix error messages. Also start a loop function when operating in UDP mode so that we can do some more stuff async. Handle element messages from udpsrc to detect timeouts. If a timeout happens we currently generate an error. API: rtspsrc::timeout property. * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_create): Really implement the timeout in microseconds and not milliseconds.
2006-09-29 15:37:29 +00:00
rtsp://streamr.hitpops.jp/ngc/mov/m0609.mov
gst/rtsp/URLS: Added some test URLS. Original commit message from CVS: * gst/rtsp/URLS: Added some test URLS. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_loop), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: When creating streams, give access to the complete SDP. Fix some leaks. Collect and merge global stream properties in stream caps. Preliminary support for WMServer. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspconnection.h: Make connection interruptable. Refactor to make it reconnectable. Don't fail on short reads when reading data packets. * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port), (rtsp_url_get_port): * gst/rtsp/rtspurl.h: Add methods for getting/setting the port. * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_message_get_attribute_val), (sdp_media_get_attribute), (sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val), (sdp_media_get_format), (sdp_parse_line), (sdp_message_parse_buffer): Fix headers. Add methods for getting multiple attributes with the same name. Increase buffer size when parsing. Fix parsing of a=foo fields. * gst/rtsp/test.c: (main): Update to new connection API. * gst/rtsp/rtspmessage.c: (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/rtsptransport.c: (rtsp_transport_free): * gst/rtsp/rtsptransport.h: * gst/rtsp/sdp.h: * gst/rtsp/sdpmessage.h: * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtsp.h: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtpdec.h: * gst/rtsp/rtsp.h: * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: Dual licensed under MIT and LGPL now.
2006-09-20 16:06:27 +00:00
ASF (audio/video):
rtsp://aod.mylisten.com/aod/8/03/069803_0903135.wma
rtsp://195.246.8.69/dur_pull_hi
rtsp://195.219.160.200/mcp?chid=25&pid=196&vid=2402201&br=800&tid=1&void=2844&muk=H1J4yFr0mJB
rtsp://a1174.v26630f.c26630.g.vm.akamaistream.net/7/1174/26630/361/od-video.msn.com/8/mbr/rs_perf_dilana_091206_finale9135.wmv
MP4V-ES/mpeg4-generic(ACC):
rtsp://vod.nwec.jp/quicktime/505.mov
gst/rtsp/URLS: Add some more URLs. Original commit message from CVS: * gst/rtsp/URLS: Add some more URLs. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_loop), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Add timeout property to control UDP timeouts. Fix error messages. Also start a loop function when operating in UDP mode so that we can do some more stuff async. Handle element messages from udpsrc to detect timeouts. If a timeout happens we currently generate an error. API: rtspsrc::timeout property. * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_create): Really implement the timeout in microseconds and not milliseconds.
2006-09-29 15:37:29 +00:00
rtsp://203.140.68.241:554/hirakataeizou9.mp4
gst/rtsp/URLS: Add example H264 rtsp url. Original commit message from CVS: * gst/rtsp/URLS: Add example H264 rtsp url. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): Don't convert values to lowercase or we might mess up base64 encoded properties.
2007-02-16 12:32:01 +00:00
rtsp://kmdi.utoronto.ca:555/osconf/2004_may9.1.mp4
gst/rtsp/URLS: Add some more URLs. Original commit message from CVS: * gst/rtsp/URLS: Add some more URLs. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_loop), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Add timeout property to control UDP timeouts. Fix error messages. Also start a loop function when operating in UDP mode so that we can do some more stuff async. Handle element messages from udpsrc to detect timeouts. If a timeout happens we currently generate an error. API: rtspsrc::timeout property. * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_create): Really implement the timeout in microseconds and not milliseconds.
2006-09-29 15:37:29 +00:00
REAL:
rtsp://213.254.239.61/farm/*/encoder/tagesschau/live1high.rm
gst/rtsp/URLS: Added some other URL. Original commit message from CVS: * gst/rtsp/URLS: Added some other URL. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp), (gst_rtspsrc_handle_request), (gst_rtspsrc_send), (gst_rtspsrc_open), (gst_rtspsrc_play), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Work on fallback to TCP connection when the UDP socket times out. Handler server requests, just reply with OK for now. * gst/rtsp/rtspdefs.c: (rtsp_strresult): * gst/rtsp/rtspdefs.h: Added some more Real extension headers. * gst/rtsp/rtspurl.c: (rtsp_url_parse): Fix parsing of urls with a ':' that is not part of the hostname:port part of the url.
2006-10-11 16:21:53 +00:00
rtsp://64.192.137.105:554/real.amazon-de.eu2/phononet/B/0/0/0/H/W/Y/4/K/S/01.01.rm?cloakport=80,554,7070
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