1. 14 May, 2007 3 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: When we try to execute a method that is not supported... · 789ef040
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.c: When we try to execute a method that is not supported by the server, don't error out but remov...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
      When we try to execute a method that is not supported by the server,
      don't error out but remove the method from the accepted methods so that
      we never try to perform this method again.
      789ef040
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Parse range correctly. · 63b73eff
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_range):
      Parse range correctly.
      * gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
      The baseurl now always has a '/' at the start.
      63b73eff
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Factor out caps configuration and configure more stuff... · fc2f6baf
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.c: Factor out caps configuration and configure more stuff such as the time ranges and speed/scale...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps),
      (gst_rtspsrc_parse_range), (gst_rtspsrc_open),
      (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
      Factor out caps configuration and configure more stuff such as the time
      ranges and speed/scale values.
      * gst/rtsp/rtsptransport.c:
      Add Copyright after non-trival fixes.
      fc2f6baf
  2. 11 May, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/: Preliminary seek support. · 02fa0a79
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
      (gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
      (gst_rtspsrc_handle_src_event),
      (gst_rtspsrc_stream_configure_manager),
      (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open),
      (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
      * gst/rtsp/gstrtspsrc.h:
      * gst/rtsp/rtspdefs.h:
      Preliminary seek support.
      Activate internal pads so that we can receive events on them.
      Don't try to parse a range string when it's NULL.
      02fa0a79
  3. 09 May, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/: Add code to parse time ranges. · d29215b2
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/Makefile.am:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_query),
      (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open):
      * gst/rtsp/rtsprange.c: (parse_npt_time), (parse_npt_range),
      (parse_clock_range), (parse_smpte_range), (rtsp_range_parse),
      (rtsp_range_free):
      * gst/rtsp/rtsprange.h:
      Add code to parse time ranges.
      Report DURATION on the stream when possible.
      d29215b2
  4. 04 May, 2007 3 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Let more error state trickle down so that we can catch more error cases. · 9e37243e
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
      (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
      (gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
      (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
      (gst_rtspsrc_change_state):
      Let more error state trickle down so that we can catch more error
      cases.
      Handle keep-alive a little smarter by selecting a method the server
      actually supports.
      Fix a race in UDP streaming shutdown.
      9e37243e
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Ignore errors when trying to use the keep-alive messages. · 5f2fbbd7
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive):
      Ignore errors when trying to use the keep-alive messages.
      5f2fbbd7
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Send RTCP messages back to the server over the TCP connection. · fb80e579
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
      (gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain),
      (gst_rtspsrc_stream_configure_manager),
      (gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
      (gst_rtspsrc_stream_configure_mcast),
      (gst_rtspsrc_stream_configure_udp),
      (gst_rtspsrc_stream_configure_udp_sink),
      (gst_rtspsrc_stream_configure_transport):
      Send RTCP messages back to the server over the TCP connection.
      * gst/rtsp/rtspconnection.c: (rtsp_connection_write),
      (rtsp_connection_send), (rtsp_connection_read), (read_body),
      (rtsp_connection_receive):
      * gst/rtsp/rtspconnection.h:
      Factor out and expose lowlevel _write and _read methods.
      Implement sending data messages to the server.
      fb80e579
  5. 03 May, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Refactor transport configuration code. · 17011e9a
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
      (gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event),
      (gst_rtspsrc_handle_src_query),
      (gst_rtspsrc_stream_configure_manager),
      (gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
      (gst_rtspsrc_stream_configure_mcast),
      (gst_rtspsrc_stream_configure_udp),
      (gst_rtspsrc_stream_configure_udp_sink),
      (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
      (gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
      (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
      (gst_rtspsrc_pause):
      Refactor transport configuration code.
      Create internal pads for TCP transport so that we can implement events
      and queries.
      Handle events and queries.
      Parse range from the SDP.
      Fix race in pause handler where the connection could still be flushing.
      17011e9a
  6. 02 May, 2007 2 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Fix race when multiple udp sources post timeouts, just... · 24e51b3c
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.*: Fix race when multiple udp sources post timeouts, just act on the first received timeout.
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
      (gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
      (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
      (gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
      (gst_rtspsrc_play), (gst_rtspsrc_handle_message),
      (gst_rtspsrc_change_state):
      * gst/rtsp/gstrtspsrc.h:
      Fix race when multiple udp sources post timeouts, just act on the first
      received timeout.
      Protect stream list with a recursive lock to fix some races.
      Flush connection when we need to do a reconnect or stop.
      Make state lock recursive.
      * gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
      (rtsp_connection_close):
      Some small cleanups.
      24e51b3c
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place. · 92396be1
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
      (gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
      (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
      (gst_rtspsrc_open), (gst_rtspsrc_handle_message):
      * gst/rtsp/gstrtspsrc.h:
      Fix sending RTCP to the right place.
      Fix bug in reffing the wrong UDP element.
      Use new pad names for the session manager.
      Implement handling server requests in interleaved and UDP modes.
      Handle session keep-alive in UDP modes.
      Remove GCond for handling UDP timeouts.
      * gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
      (rtsp_connection_send), (rtsp_connection_read), (read_body),
      (rtsp_connection_receive), (rtsp_connection_close):
      * gst/rtsp/rtspconnection.h:
      Store connection IP address for later.
      Add timeout args to all operations that might block forever.
      Parse session timeout.
      Only close sockets when not already closed.
      * gst/rtsp/rtspdefs.c:
      * gst/rtsp/rtspdefs.h:
      Add timeout return value and error string.
      * gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
      Add small comment.
      92396be1
  7. 29 Apr, 2007 1 commit
    • Wim Taymans's avatar
      gst/udp/gstmultiudpsink.c: Add code to drop membership of a multicast group. · 066598d8
      Wim Taymans authored
      Original commit message from CVS:
      * gst/udp/gstmultiudpsink.c: (leave_multicast),
      (gst_multiudpsink_add), (gst_multiudpsink_remove):
      Add code to drop membership of a multicast group.
      * gst/udp/gstudpsink.c: (gst_udpsink_update_uri),
      (gst_udpsink_set_uri):
      Implement URI handler.
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_parse_rtpinfo):
      Use URI handler to make udpsink instace.
      Improve code to configure port and destination.
      066598d8
  8. 27 Apr, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Ignore ASYNC state messages from the udpsink, it's... · 6a790cb7
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.c: Ignore ASYNC state messages from the udpsink, it's irrelevant for the parent.
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
      (gst_rtspsrc_handle_message):
      Ignore ASYNC state messages from the udpsink, it's irrelevant for the
      parent.
      6a790cb7
  9. 26 Apr, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Protect state changes with a lock. · 530f214b
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
      (gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_open), (gst_rtspsrc_close),
      (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
      (gst_rtspsrc_pause):
      * gst/rtsp/gstrtspsrc.h:
      Protect state changes with a lock.
      * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
      (parse_line):
      * gst/rtsp/rtspconnection.h:
      Remove some unused stuff.
      530f214b
  10. 25 Apr, 2007 2 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtpdec.*: Add dummy latency property to be backwards compat with rtpbin. · 6937be1a
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init),
      (gst_rtp_dec_init), (gst_rtp_dec_set_property),
      (gst_rtp_dec_get_property):
      * gst/rtsp/gstrtpdec.h:
      Add dummy latency property to be backwards compat with rtpbin.
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
      (gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
      (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_parse_rtpinfo):
      * gst/rtsp/gstrtspsrc.h:
      Add latency property and configure in the session manager.
      Don't set invalid clock-base and seqnum-base on caps, some servers
      sometimes don't send them.
      6937be1a
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Parse server address from SDP. · 1beeda3f
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
      (gst_rtspsrc_stream_free), (request_pt_map),
      (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open):
      * gst/rtsp/gstrtspsrc.h:
      Parse server address from SDP.
      Hook up a udpsink to send RTCP back to the server.
      * docs/plugins/gst-plugins-good-plugins-sections.txt:
      * gst/rtsp/rtsptransport.h:
      Add some docs.
      1beeda3f
  11. 13 Apr, 2007 1 commit
    • Wim Taymans's avatar
      docs/plugins/gst-plugins-good-plugins-sections.txt: Fix docs. · b7524708
      Wim Taymans authored
      Original commit message from CVS:
      * docs/plugins/gst-plugins-good-plugins-sections.txt:
      Fix docs.
      * gst/rtsp/URLS:
      Add some more example urls.
      * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
      (gst_rtp_dec_chain_rtp):
      Better debugging.
      * gst/rtsp/gstrtspsrc.c: (request_pt_map),
      (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_parse_rtpinfo):
      Remove unused code.
      b7524708
  12. 12 Apr, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtpdec.*: Make backward compat with rtpbin by adding the request-pt-map signals. · 86a4c1c6
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
      (gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
      * gst/rtsp/gstrtpdec.h:
      Make backward compat with rtpbin by adding the request-pt-map signals.
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
      (new_session_pad), (request_pt_map),
      (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_stream_configure_caps),
      (gst_rtspsrc_activate_streams):
      * gst/rtsp/gstrtspsrc.h:
      Implement request-pt-map signals instead of setting caps on the buffers
      for the session manager.
      86a4c1c6
  13. 10 Apr, 2007 1 commit
    • Peter Kjellerstedt's avatar
      gst/: Fix some compiler warnings. Fixes #428182. · 50f88db3
      Peter Kjellerstedt authored and Wim Taymans's avatar Wim Taymans committed
      Original commit message from CVS:
      Patch by: Peter Kjellerstedt  <pkj at axis com>
      * gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
      * gst/rtp/gstrtpL16depay.c:
      * gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
      * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode),
      (gst_rtp_speex_depay_setcaps):
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
      (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp):
      * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send):
      Fix some compiler warnings. Fixes #428182.
      50f88db3
  14. 06 Apr, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback. · f80444aa
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/Makefile.am:
      * gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
      (free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
      (gst_rtp_dec_init), (gst_rtp_dec_finalize),
      (gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
      (gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
      (gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
      (gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
      (create_rtcp), (gst_rtp_dec_request_new_pad),
      (gst_rtp_dec_release_pad):
      * gst/rtsp/gstrtpdec.h:
      * gst/rtsp/gstrtsp.c: (plugin_init):
      Morph RTPDec into something compatible with RTPBin as a fallback.
      Various other style fixes.
      * gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
      (find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
      (gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
      (new_session_pad), (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
      (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
      * gst/rtsp/gstrtspsrc.h:
      Implement RTPBin session manager handling.
      Don't try to add empty properties to caps.
      Implement fallback session manager, handling.
      Don't combine errors from RTCP streams, just ignore them.
      * gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
      * gst/rtsp/rtsptransport.h:
      Implement fallback session manager.
      Make RTPBin the default one when available.
      f80444aa
  15. 25 Mar, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types,... · 8f5fb88b
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types, rearrange stuff so that the rtpmap field ...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),
      (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free),
      (get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap),
      (gst_rtspsrc_media_to_caps),
      (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_stream_configure_caps),
      (gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo):
      * gst/rtsp/gstrtspsrc.h:
      Handle default clock-rates for static payload types, rearrange stuff so
      that the rtpmap field in the sdp can override the defaults.
      Parse RTP-Info field to get the seqnum and timebase fields that should
      go in the caps.
      Delay configuring caps after we got the RTP-Info from the PLAY reply from
      the server.
      8f5fb88b
  16. 09 Mar, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: When activated, remove the udpsrc timeout, we have... · beef8e01
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.c: When activated, remove the udpsrc timeout, we have dataflow and timeouts will later be handled...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
      When activated, remove the udpsrc timeout, we have dataflow and timeouts
      will later be handled by the jitterbuffer.
      beef8e01
  17. 04 Mar, 2007 1 commit
    • Jan Schmidt's avatar
      Fix a bunch of leaks shown by the newly-added states test. · de1357a4
      Jan Schmidt authored
      Original commit message from CVS:
      * ext/flac/gstflacenc.c: (gst_flac_enc_finalize):
      * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init),
      (gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize):
      * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init),
      (gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose),
      (gst_gconf_audio_src_finalize), (do_toggle_element):
      * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init),
      (gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize),
      (do_toggle_element):
      * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init),
      (gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose),
      (gst_gconf_video_src_finalize), (do_toggle_element):
      * ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init),
      (gst_switch_sink_reset), (gst_switch_sink_set_child):
      * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
      * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
      * ext/shout2/gstshout2.c: (gst_shout2send_class_init),
      (gst_shout2send_init), (gst_shout2send_finalize):
      * gst/debug/testplugin.c: (gst_test_class_init),
      (gst_test_finalize):
      * gst/flx/gstflxdec.c: (gst_flxdec_class_init),
      (gst_flxdec_dispose):
      * gst/multipart/multipartmux.c: (gst_multipart_mux_finalize):
      * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize):
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
      (gst_rtspsrc_finalize):
      * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context):
      * gst/rtsp/rtspextwms.h:
      * gst/smpte/gstsmpte.c: (gst_smpte_class_init),
      (gst_smpte_finalize):
      * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize):
      * gst/udp/gstudpsink.c: (gst_udpsink_class_init),
      (gst_udpsink_finalize):
      * gst/wavparse/gstwavparse.c: (gst_wavparse_dispose),
      (gst_wavparse_sink_activate):
      * sys/oss/gstosssink.c: (gst_oss_sink_finalise):
      * sys/oss/gstosssrc.c: (gst_oss_src_class_init),
      (gst_oss_src_finalize):
      * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy):
      * sys/v4l2/gstv4l2object.h:
      * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
      (gst_v4l2src_finalize):
      * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):
      Fix a bunch of leaks shown by the newly-added states test.
      de1357a4
  18. 01 Mar, 2007 2 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Errors from the udp sources are not fatal unless all of them are in error. · 84c6cb98
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (find_stream_by_channel),
      (find_stream_by_udpsrc), (gst_rtspsrc_handle_message):
      Errors from the udp sources are not fatal unless all of them are in
      error.
      84c6cb98
    • Wim Taymans's avatar
      gst/rtsp/gstrtpdec.c: Make state change to PAUSED NO_PREROLL because that's... · dc212cdb
      Wim Taymans authored
      gst/rtsp/gstrtpdec.c: Make state change to PAUSED NO_PREROLL because that's what it will be in the future and rtspsrc...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state):
      Make state change to PAUSED NO_PREROLL because that's what it will be in
      the future and rtspsrc relies on it.
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_change_state):
      Don't error out when we don't get an error from the state change
      function.
      dc212cdb
  19. 23 Feb, 2007 1 commit
    • Jan Schmidt's avatar
      gst/rtsp/: Implement simple Basic Authentication support so that urls like... · 66df66da
      Jan Schmidt authored
      gst/rtsp/: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work ...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
      (gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
      (gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
      (gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
      (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
      (gst_rtspsrc_uri_set_uri):
      * gst/rtsp/gstrtspsrc.h:
      * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
      (append_auth_header), (rtsp_connection_send),
      (rtsp_connection_free), (rtsp_connection_set_auth):
      * gst/rtsp/rtspconnection.h:
      * gst/rtsp/rtspdefs.h:
      * gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
      * gst/rtsp/rtspurl.h:
      Implement simple Basic Authentication support so that urls like
      rtsp://user:pass@hostname/rtspstream work on hosts that require
      authentication.
      66df66da
  20. 16 Feb, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/URLS: Add example H264 rtsp url. · 7fd02504
      Wim Taymans authored
      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.
      7fd02504
  21. 11 Feb, 2007 1 commit
    • Sebastien Moutte's avatar
      gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 seems to do not support it. · 9c8ea356
      Sebastien Moutte authored
      Original commit message from CVS:
      * gst/avi/gstavimux.c:
      Comment a #if 0 in caps template definition as VS6 seems to
      do not support it.
      * gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp):
      Use gst_guint64_to_gdouble for conversion.
      * gst/rtsp/rtspconnection.c:(rtsp_connection_send):
      Move variables declaration before the first instruction.
      * gst/rtsp/rtspdefs.c:(rtsp_strresult):
      Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported.
      And don't include netdb.h for G_OS_WIN32
      * gst/rtsp/sdpmessage.c:(sdp_parse_line):
      This initialization SDPMedia nmedia = {.media = NULL }; is not supported
      by VS6 then use an other way to initialize SDPMedia structure.
      * gst/udp/gstdynudpsink.h:
      * gst/udp/gstdynudpnetutils.h:
      Do not include <sys/time.h> for G_OS_WIN32
      * gst/udp/gstudpsrc.c:
      Define socklen_t as int for G_OS_WIN32
      * win/common/config.h.in:
      Undef HAVE_NETINET_IN_H
      * win32/vs6/gst_plugins_good.dsw:
      * win32/vs6/libgstrtp.dsp:
      * win32/vs6/libgstrtsp.dsp:
      * win32/vs6/libgstautogen.dsp:
      * win32/vs6/libgstaudiofx.dsp:
      * win32/vs6/libgstudp.dsp:
      Add and update project files.
      * win32/common/gstudp-enumtypes.c:
      * win32/common/gstudp-enumtypes.h:
      Add a copy of udp enumtypes to win32/common as in core
      and base.
      9c8ea356
  22. 25 Jan, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Convert SDP fields to upper/lowercase following the... · 2de7376a
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.c: Convert SDP fields to upper/lowercase following the rules in the SDP to caps document.
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
      (gst_rtspsrc_activate_streams):
      Convert SDP fields to upper/lowercase following the rules in the SDP to
      caps document.
      2de7376a
  23. 24 Jan, 2007 2 commits
  24. 11 Jan, 2007 1 commit
  25. 10 Jan, 2007 1 commit
    • Peter Kjellerstedt's avatar
      gst/rtsp/: Allow url to be NULL to be able to use it for server connections. · 12ab127d
      Peter Kjellerstedt authored and Wim Taymans's avatar Wim Taymans committed
      Original commit message from CVS:
      Patch by: Peter Kjellerstedt  <pkj at axis com>
      * gst/rtsp/COPYING.MIT:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
      (gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup),
      (gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked),
      (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_loop_udp), (gst_rtspsrc_send),
      (gst_rtspsrc_parse_methods),
      (gst_rtspsrc_create_transports_string),
      (gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
      (gst_rtspsrc_open), (gst_rtspsrc_close):
      * gst/rtsp/gstrtspsrc.h:
      * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
      (rtsp_connection_connect), (rtsp_connection_send), (read_line),
      (parse_request_line), (parse_line), (rtsp_connection_read),
      (rtsp_connection_close):
      * gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult),
      (rtsp_method_as_text), (rtsp_header_as_text),
      (rtsp_status_as_text), (rtsp_find_header_field),
      (rtsp_find_method):
      * gst/rtsp/rtspdefs.h:
      * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send),
      (rtsp_ext_wms_configure_stream):
      * gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init),
      (rtsp_message_new_request), (rtsp_message_init_request),
      (rtsp_message_new_response), (rtsp_message_init_response),
      (rtsp_message_init_data), (rtsp_message_unset),
      (rtsp_message_free), (rtsp_message_add_header),
      (rtsp_message_get_header), (rtsp_message_set_body),
      (rtsp_message_get_body), (dump_mem), (rtsp_message_dump):
      * gst/rtsp/rtspmessage.h:
      * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
      (sdp_media_get_attribute_val_n), (read_string), (read_string_del),
      (sdp_parse_line), (sdp_message_parse_buffer), (print_media),
      (sdp_message_dump):
      Allow url to be NULL to be able to use it for server connections.
      Can now send responses as well as requests.
      No longer hangs in an endless loop if EOF is received.
      Can now convert a status code to a text string.
      Return RTSP_HDR_INVALID for unknown headers.
      Return RTSP_INVALID for unknown methods.
      Copy CSeq and Session headers from the request.
      Only free memory corresponding to the currently set message type.
      Added const to function arguments as appropriate.
      Avoid a compiler warning when initializing nmedia.
      Use guint rather than gint to avoid compiler warnings.
      Fix crasher in wms extension.
      Factor out stream setup from open_connection.
      Delay activation of streams when actual data is received from the
      server, this prepares us to do proper protocol switching.
      Added new license.
      Fixes #380895.
      12ab127d
  26. 28 Nov, 2006 1 commit
    • Wim Taymans's avatar
      gst/rtsp/: Add method so that extensions can choose to disable the setup of a stream. · f249d639
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
      * gst/rtsp/gstrtspsrc.h:
      * gst/rtsp/rtspext.h:
      * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream),
      (rtsp_ext_wms_get_context):
      Add method so that extensions can choose to disable the setup of
      a stream.
      Make the WMS extension skip setup of x-wms-rtx streams. Fixes #377792.
      f249d639
  27. 18 Oct, 2006 1 commit
    • Wim Taymans's avatar
      gst/rtsp/: Reuse already existing enum for lower transport. · b14738fb
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
      (gst_rtspsrc_class_init), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
      (gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_set_uri):
      * gst/rtsp/rtspconnection.c: (rtsp_connection_create):
      * gst/rtsp/rtspdefs.h:
      * gst/rtsp/rtspurl.c: (rtsp_url_parse):
      * gst/rtsp/rtspurl.h:
      Reuse already existing enum for lower transport.
      Add rtspt and rtspu protocols.
      Send redirect to rtspt when udp times out.
      b14738fb
  28. 16 Oct, 2006 1 commit
    • Josep Torra Valles's avatar
      Fix a bunch of problems discovered by the Forte compiler, mostly type mixups... · c4e7ebfe
      Josep Torra Valles authored and Tim-Philipp Müller's avatar Tim-Philipp Müller committed
      Fix a bunch of problems discovered by the Forte compiler, mostly type mixups and pointer arithmetics with void pointe...
      
      Original commit message from CVS:
      Patch by: Josep Torra Valles  <josep at fluendo com>
      * ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
      * ext/esd/esdsink.c: (gst_esdsink_write):
      * ext/flac/gstflacdec.c: (gst_flac_dec_length),
      (gst_flac_dec_read_seekable), (gst_flac_dec_chain),
      (gst_flac_dec_send_newsegment):
      * ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback),
      (gst_flac_enc_tell_callback):
      * ext/jpeg/smokecodec.c: (find_best_size), (smokecodec_encode),
      (smokecodec_parse_header), (smokecodec_decode):
      * gst/avi/gstavimux.c: (gst_avi_mux_write_avix_index):
      * gst/debug/efence.c: (gst_fenced_buffer_alloc):
      * gst/goom/Makefile.am:
      * gst/goom/gstgoom.c:
      * gst/icydemux/gsticydemux.c: (gst_icydemux_typefind_or_forward):
      * gst/rtsp/gstrtspsrc.c:
      * gst/rtsp/rtspconnection.c: (rtsp_connection_read):
      * gst/udp/gstudpsink.c:
      * gst/udp/gstudpsrc.c:
      * gst/wavparse/gstwavparse.c: (gst_wavparse_change_state):
      * sys/sunaudio/gstsunaudiomixertrack.h:
      Fix a bunch of problems discovered by the Forte compiler, mostly type
      mixups and pointer arithmetics with void pointers. Fixes #362603.
      c4e7ebfe
  29. 11 Oct, 2006 1 commit
    • Wim Taymans's avatar
      gst/rtsp/URLS: Added some other URL. · 7accf76d
      Wim Taymans authored
      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.
      7accf76d
  30. 07 Oct, 2006 1 commit
  31. 06 Oct, 2006 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Rework how the transport string is constructed, try to... · a600d311
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.*: Rework how the transport string is constructed, try to share channels and udp ports.
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
      (gst_rtspsrc_class_init), (gst_rtspsrc_init),
      (gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps),
      (gst_rtspsrc_alloc_udp_ports),
      (gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
      (gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_create_transports_string),
      (gst_rtspsrc_configure_transports), (gst_rtspsrc_open),
      (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
      * gst/rtsp/gstrtspsrc.h:
      Rework how the transport string is constructed, try to share channels
      and udp ports.
      Make most of the stuff less dependant on RTP as we are also going to use
      it for RDT.
      Add support for transport specific session managers.
      * gst/rtsp/rtspconnection.c: (rtsp_connection_flush):
      Implement _flush().
      * gst/rtsp/rtspdefs.c: (rtsp_strresult):
      * gst/rtsp/rtspdefs.h:
      Add generic error return code.
      * gst/rtsp/rtspext.h:
      Add support for pluggable tranport strings.
      * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send),
      (rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp),
      (rtsp_ext_wms_get_context):
      Detect WMServer and activate the extension.
      * gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime),
      (rtsp_transport_get_manager), (rtsp_transport_parse):
      * gst/rtsp/rtsptransport.h:
      Added methods to get mime/manager for certain transports.
      a600d311
  32. 05 Oct, 2006 1 commit
    • Tim-Philipp Müller's avatar
      Printf format fixes. · 82f5a350
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * ext/cairo/gsttimeoverlay.c:
      (gst_cairo_time_overlay_update_font_height):
      * ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_transform_caps):
      * ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_parse_image_data):
      * ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
      * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
      * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
      * ext/libpng/gstpngdec.c: (user_endrow_callback):
      * gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
      * gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex),
      (gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
      (gst_avi_demux_stream_data):
      * gst/cutter/gstcutter.c: (gst_cutter_chain):
      * gst/debug/efence.c: (gst_efence_buffer_alloc),
      (gst_fenced_buffer_copy):
      * gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
      * gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
      * gst/matroska/matroska-mux.c: (gst_matroska_mux_start):
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
      (gst_rtspsrc_handle_message):
      * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
      * sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
      Printf format fixes.
      82f5a350