1. 14 Nov, 2007 1 commit
  2. 22 Oct, 2007 1 commit
  3. 17 Oct, 2007 1 commit
  4. 08 Oct, 2007 2 commits
    • Jan Schmidt's avatar
      gst/rtsp/gstrtspsrc.c: Fix compiler warning by using GST_CLOCK_TIME_NONE to... · 3ca2d477
      Jan Schmidt authored
      gst/rtsp/gstrtspsrc.c: Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise a GstClockTime.
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush):
      Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise
      a GstClockTime.
      3ca2d477
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: More seeking fixes, mostly passing around the new... · 92e16a65
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.c: More seeking fixes, mostly passing around the new playback segment in order to configure it pr...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
      (gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
      (gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp),
      (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play),
      (gst_rtspsrc_change_state):
      More seeking fixes, mostly passing around the new playback segment in
      order to configure it properly.
      Also reset base_time of udp sources when setting them back to PLAYING as
      a temporary hack until core supports seek in live sources properly.
      92e16a65
  5. 05 Oct, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Improve flushing behaviour. · 7624f914
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
      (gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event),
      (gst_rtspsrc_handle_internal_src_query),
      (gst_rtspsrc_handle_src_query), (new_session_pad),
      (gst_rtspsrc_stream_configure_tcp),
      (gst_rtspsrc_stream_configure_transport),
      (gst_rtspsrc_loop_send_cmd):
      Improve flushing behaviour.
      Set state of the udp sources to PAUSE/PLAYING correctly.
      Handle events and queries for UDP and TCP transport now.
      7624f914
  6. 01 Oct, 2007 2 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Parse bandwidth modifiers, they are not yet configured... · 5274c3f4
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.*: Parse bandwidth modifiers, they are not yet configured in the session manager because we don't...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth),
      (gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream),
      (gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
      * gst/rtsp/gstrtspsrc.h:
      Parse bandwidth modifiers, they are not yet configured in the session
      manager because we don't have an API for that yet.
      5274c3f4
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Use shiny new function in -base to get the default clock-rate. · b3e03a9a
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
      (gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
      Use shiny new function in -base to get the default clock-rate.
      Update some docs.
      b3e03a9a
  7. 28 Sep, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: In TCP mode, only timestamp the first buffer. TCP is... · bea90106
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.*: In TCP mode, only timestamp the first buffer. TCP is not real time and it does not make sense ...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_play):
      * gst/rtsp/gstrtspsrc.h:
      In TCP mode, only timestamp the first buffer. TCP is not real time and
      it does not make sense to try to skew compensate, also some servers send
      the first batch of data in a burst.
      bea90106
  8. 26 Sep, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Set timestamps on RTP buffers in interleaved mode. · 4683ff80
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
      (gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_loop_udp), (gst_rtspsrc_close):
      * gst/rtsp/gstrtspsrc.h:
      Set timestamps on RTP buffers in interleaved mode.
      Mark first buffers with a DISCONT.
      Remove flush hack now that sync for live sources has been figured out.
      4683ff80
  9. 17 Sep, 2007 2 commits
  10. 29 Aug, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Use new basesink async property to make sparse RTCP... · 14e218c0
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.c: Use new basesink async property to make sparse RTCP packet not wait for preroll.
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
      (gst_rtspsrc_dup_printf):
      Use new basesink async property to make sparse RTCP packet not wait for
      preroll.
      14e218c0
  11. 23 Aug, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Make sure we generate and parse floating point values... · a221e919
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.c: Make sure we generate and parse floating point values in the POSIX locale instead of the curre...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf),
      (gst_rtspsrc_get_float), (gst_rtspsrc_play):
      Make sure we generate and parse floating point values in the POSIX
      locale instead of the current locale.
      a221e919
  12. 22 Aug, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Fix method detection again. · 5592bdd4
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
      (gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
      (gst_rtspsrc_play):
      * gst/rtsp/gstrtspsrc.h:
      Fix method detection again.
      Keep track of when we must send a Range header.
      Use segment values for Range, Speed and Scale headers.
      Parse Speed and Scale headers to update the segment values.
      5592bdd4
  13. 18 Aug, 2007 1 commit
  14. 17 Aug, 2007 3 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Protect connection activity with a new lock, avoids... · 0dcafb06
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.*: Protect connection activity with a new lock, avoids deadlocks when going to PAUSED. Fixes #455...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
      (gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
      (gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
      (gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
      (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
      (gst_rtspsrc_try_send), (gst_rtspsrc_pause):
      * gst/rtsp/gstrtspsrc.h:
      Protect connection activity with a new lock, avoids deadlocks when going
      to PAUSED. Fixes #455808.
      0dcafb06
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Fix stray %u in debug line as spotted by Saur on IRC. · 98fb7c07
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_stream_eos):
      Fix stray %u in debug line as spotted by Saur on IRC.
      98fb7c07
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Improve timeout handling. · 6ef70550
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
      (gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
      (gst_rtspsrc_stream_configure_udp_sink),
      (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
      (gst_rtspsrc_try_send), (gst_rtspsrc_send),
      (gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
      (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
      (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
      * gst/rtsp/gstrtspsrc.h:
      Improve timeout handling.
      Use the same socket for sending and receiving RTCP packets so that some
      servers can track clients better.
      Improve connection closed handling. Try to reconnect.
      Don't overwrite our content base with NULL.
      Improve debugging.
      Improve range parsing and handling.
      Remove flushing hack now that core does the right thing.
      6ef70550
  15. 16 Aug, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtpdec.*: Add (dummy) SSRC management signals. · 41f04967
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
      (gst_rtp_dec_class_init):
      * gst/rtsp/gstrtpdec.h:
      Add (dummy) SSRC management signals.
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
      (gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
      (find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
      (request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
      (on_timeout), (gst_rtspsrc_stream_configure_manager),
      (gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
      (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
      (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
      * gst/rtsp/gstrtspsrc.h:
      Add connection-speed property.
      Add find_stream helper functions.
      Handle stream EOS based on BYE messages or SSRC timeout.
      Returns SUCCESS from the state change function as we hide our async
      elements from the parent.
      41f04967
  16. 03 Aug, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Fix default clock-rate for realmedia. · a654ab9f
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (get_default_rate_for_pt),
      (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
      (gst_rtspsrc_stream_configure_tcp),
      (gst_rtspsrc_stream_configure_udp_sink):
      Fix default clock-rate for realmedia.
      Fix parsing of transport.
      Don't try to link NULL pads.
      a654ab9f
  17. 27 Jul, 2007 2 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: If we don't hav a session manager, set the caps on... · 9ace6772
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.c: If we don't hav a session manager, set the caps on outgoing buffers ourselves.
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams),
      (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_methods),
      (gst_rtspsrc_create_transports_string),
      (gst_rtspsrc_prepare_transports):
      If we don't hav a session manager, set the caps on outgoing buffers
      ourselves.
      Force PAUSE/PLAY methods for now until the extensions can overwrite.
      Append final bit of the transport string even when it does not contain a
      placeholder.
      9ace6772
    • Wim Taymans's avatar
      gst/rtsp/: Clean up the interface list. · a8ee445d
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_free),
      (gst_rtsp_ext_list_connect):
      * gst/rtsp/gstrtspext.h:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
      (gst_rtspsrc_finalize), (gst_rtspsrc_send_cb):
      Clean up the interface list.
      Allow connecting to interface signals for the extensions.
      Remove old extension code.
      Free list on cleanup.
      Allow extensions to send additional RTSP messages.
      a8ee445d
  18. 26 Jul, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/: Use rank to filter out extensions. · 9fa21084
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
      (gst_rtsp_ext_list_stream_select):
      * gst/rtsp/gstrtspext.h:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
      Use rank to filter out extensions.
      Add url to stream_select interface call.
      9fa21084
  19. 25 Jul, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/: Use shiny new RTSP and SDP library. · fa9c47f1
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/Makefile.am:
      * gst/rtsp/base64.c:
      * gst/rtsp/base64.h:
      * gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
      (gst_rtsp_ext_list_init), (gst_rtsp_ext_list_get),
      (gst_rtsp_ext_list_detect_server), (gst_rtsp_ext_list_before_send),
      (gst_rtsp_ext_list_after_send), (gst_rtsp_ext_list_parse_sdp),
      (gst_rtsp_ext_list_setup_media),
      (gst_rtsp_ext_list_configure_stream),
      (gst_rtsp_ext_list_get_transports),
      (gst_rtsp_ext_list_stream_select):
      * gst/rtsp/gstrtspext.h:
      * gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
      (gst_rtspsrc_class_init), (gst_rtspsrc_init),
      (gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
      (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
      (gst_rtspsrc_flush), (gst_rtspsrc_do_seek),
      (gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager),
      (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_handle_request), (gst_rtspsrc_send_keep_alive),
      (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
      (gst_rtspsrc_loop_send_cmd), (gst_rtsp_auth_method_to_string),
      (gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
      (gst_rtspsrc_try_send), (gst_rtspsrc_send),
      (gst_rtspsrc_parse_methods),
      (gst_rtspsrc_create_transports_string),
      (gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
      (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close),
      (gst_rtspsrc_play), (gst_rtspsrc_pause),
      (gst_rtspsrc_change_state), (gst_rtspsrc_uri_set_uri):
      * gst/rtsp/gstrtspsrc.h:
      * gst/rtsp/rtsp.h:
      * gst/rtsp/rtspconnection.c:
      * gst/rtsp/rtspconnection.h:
      * gst/rtsp/rtspdefs.c:
      * gst/rtsp/rtspdefs.h:
      * gst/rtsp/rtspext.h:
      * gst/rtsp/rtspextwms.c:
      * gst/rtsp/rtspextwms.h:
      * gst/rtsp/rtspmessage.c:
      * gst/rtsp/rtspmessage.h:
      * gst/rtsp/rtsprange.c:
      * gst/rtsp/rtsprange.h:
      * gst/rtsp/rtsptransport.c:
      * gst/rtsp/rtsptransport.h:
      * gst/rtsp/rtspurl.c:
      * gst/rtsp/rtspurl.h:
      * gst/rtsp/sdp.h:
      * gst/rtsp/sdpmessage.c:
      * gst/rtsp/sdpmessage.h:
      * gst/rtsp/test.c:
      Use shiny new RTSP and SDP library.
      Implement RTSP extensions using the new interface.
      Remove a lot of old code.
      fa9c47f1
  20. 27 Jun, 2007 2 commits
  21. 24 May, 2007 2 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Init value to avoid infinte loops. · 587d2092
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods):
      Init value to avoid infinte loops.
      587d2092
    • Peter Kjellerstedt's avatar
      gst/rtsp/: Fix for new API. · 77cc870b
      Peter Kjellerstedt authored
      Original commit message from CVS:
      Patch by: Peter Kjellerstedt  <pkj at axis com>
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth),
      (gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods),
      (gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
      (gst_rtspsrc_play):
      (rtsp_connection_send), (rtsp_connection_receive):
      * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send):
      Fix for new API.
      * gst/rtsp/rtspconnection.c: (add_auth_header),
      Only add authorisation and session headers when sending messages.
      * gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init),
      (rtsp_message_init_request), (rtsp_message_init_response),
      (rtsp_message_unset), (rtsp_message_add_header),
      (rtsp_message_remove_header), (rtsp_message_get_header),
      (rtsp_message_append_headers), (dump_key_value),
      (rtsp_message_dump):
      * gst/rtsp/rtspmessage.h:
      Add support for multiple headers of the same type by storing the parsed
      headers in a GArray instaed of a hashtable.
      77cc870b
  22. 20 May, 2007 1 commit
  23. 18 May, 2007 3 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Don't crash when an unsupported transport error was... · fc99abef
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.c: Don't crash when an unsupported transport error was returned by the server, just try to config...
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
      Don't crash when an unsupported transport error was returned by the
      server, just try to configure the next stream. Fixes #439255.
      fc99abef
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.*: Add TCP timeout property and use it for all TCP connection. · e04f7a82
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
      (gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
      (gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved),
      (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
      (gst_rtspsrc_setup_streams), (gst_rtspsrc_open):
      * gst/rtsp/gstrtspsrc.h:
      Add TCP timeout property and use it for all TCP connection.
      * gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
      (rtsp_connection_write), (rtsp_connection_next_timeout),
      (rtsp_connection_reset_timeout):
      Make connect and writes cancelable and make them use the timeout.
      e04f7a82
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Refactor timeout handling. · e4720e28
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
      (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
      (gst_rtspsrc_try_send), (gst_rtspsrc_send),
      (gst_rtspsrc_setup_streams):
      Refactor timeout handling.
      Also send keep-alive when dealing with TCP transport.
      * gst/rtsp/rtspconnection.c: (rtsp_connection_create),
      (rtsp_connection_free), (rtsp_connection_next_timeout),
      (rtsp_connection_reset_timeout):
      * gst/rtsp/rtspconnection.h:
      Use a timer to handle the session timeouts, add some methods to deal
      with timeouts.
      e4720e28
  24. 17 May, 2007 1 commit
    • Wim Taymans's avatar
      gst/rtsp/gstrtspsrc.c: Ignore streams that fail the setup command, we will... · ccd7a136
      Wim Taymans authored
      gst/rtsp/gstrtspsrc.c: Ignore streams that fail the setup command, we will retry with a different transport later on.
      
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
      (gst_rtspsrc_setup_streams):
      Ignore streams that fail the setup command, we will retry with a
      different transport later on.
      * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
      (rtsp_ext_wms_configure_stream):
      Fix encoding name case.
      ccd7a136
  25. 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
  26. 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
  27. 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
  28. 04 May, 2007 1 commit
    • 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