1. 21 May, 2007 4 commits
    • Wim Taymans's avatar
      gst/rtsp/gstrtpdec.*: Added signal for backwards compat. · 321a79d4
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init):
      * gst/rtsp/gstrtpdec.h:
      Added signal for backwards compat.
      321a79d4
    • René Stadler's avatar
      Use audioconvert for converting from non-native endianness floats in auparse... · 4bd11406
      René Stadler authored and Sebastian Dröge's avatar Sebastian Dröge committed
      Use audioconvert for converting from non-native endianness floats in auparse instead of doing it ourself. Fixes #424527.
      
      Original commit message from CVS:
      Patch by: René Stadler <mail at renestadler dot de>
      * configure.ac:
      * gst/auparse/gstauparse.c: (gst_au_parse_reset),
      (gst_au_parse_parse_header), (gst_au_parse_chain):
      * gst/auparse/gstauparse.h:
      Use audioconvert for converting from non-native endianness floats
      in auparse instead of doing it ourself. Fixes #424527.
      This needs the audioconvert from plugins-base CVS.
      4bd11406
    • Wim Taymans's avatar
      gst/rtp/gstrtph263ppay.c: Fix enum registration. · 20dc422e
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
      (gst_rtp_h263p_pay_flush):
      Fix enum registration.
      20dc422e
    • Antoine Tremblay's avatar
      gst/rtp/gstrtph263ppay.*: Add new fragmentation mode base on GOB headers. Fixes #438940. · 0ff05f81
      Antoine Tremblay authored and Wim Taymans's avatar Wim Taymans committed
      Original commit message from CVS:
      Patch by: Antoine Tremblay <hexa00 at gmail dot com>
      * gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
      (gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_init),
      (gst_rtp_h263p_pay_set_property), (gst_rtp_h263p_pay_get_property),
      (gst_rtp_h263p_pay_flush):
      * gst/rtp/gstrtph263ppay.h:
      Add new fragmentation mode base on GOB headers. Fixes #438940.
      0ff05f81
  2. 20 May, 2007 4 commits
    • Sebastian Dröge's avatar
      ext/wavpack/gstwavpackenc.c: Add missing audioconverts in the example... · d6a28f9e
      Sebastian Dröge authored
      ext/wavpack/gstwavpackenc.c: Add missing audioconverts in the example pipelines of wavpackenc. As the wavpack stuff n...
      
      Original commit message from CVS:
      * ext/wavpack/gstwavpackenc.c:
      Add missing audioconverts in the example pipelines of wavpackenc. As
      the wavpack stuff now needs input with 32 bit width (and random depth)
      this is needed now. The example pipelines for the parser and decoder
      are still fine.
      d6a28f9e
    • Tim-Philipp Müller's avatar
      sys/directdraw/gstdirectdrawsink.c: Bunch of small fixes: remove static... · fa055153
      Tim-Philipp Müller authored
      sys/directdraw/gstdirectdrawsink.c: Bunch of small fixes: remove static function that doesn't exist; declare another ...
      
      Original commit message from CVS:
      * sys/directdraw/gstdirectdrawsink.c: (gst_ddrawsurface_finalize),
      (gst_directdraw_sink_buffer_alloc),
      (gst_directdraw_sink_get_ddrawcaps),
      (gst_directdraw_sink_surface_create):
      Bunch of small fixes: remove static function that doesn't exist;
      declare another one that does; printf format fix; use right macro
      when specifying debug category; remove a bunch of unused variables;
      #if 0 out an unused chunk of code (partially fixes #439914).
      fa055153
    • Tim-Philipp Müller's avatar
      gst/: Printf format fixes (#439910, #439911). · 798b7863
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample):
      * gst/switch/gstswitch.c: (gst_switch_chain):
      Printf format fixes (#439910, #439911).
      798b7863
    • Tim-Philipp Müller's avatar
      gst/rtsp/gstrtspsrc.c: Printf format fix. · 263e0458
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
      Printf format fix.
      263e0458
  3. 19 May, 2007 1 commit
    • René Stadler's avatar
      Add replaygain playback elements (#412710). · 4e45e0a2
      René Stadler authored and Tim-Philipp Müller's avatar Tim-Philipp Müller committed
      Original commit message from CVS:
      Patch by: René Stadler <mail at renestadler de>
      * docs/plugins/Makefile.am:
      * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
      * docs/plugins/gst-plugins-bad-plugins-sections.txt:
      * docs/plugins/inspect/plugin-replaygain.xml:
      * gst/replaygain/Makefile.am:
      * gst/replaygain/gstrganalysis.c: (gst_rg_analysis_class_init),
      (gst_rg_analysis_start), (gst_rg_analysis_set_caps),
      (gst_rg_analysis_transform_ip), (gst_rg_analysis_event),
      (gst_rg_analysis_stop), (gst_rg_analysis_handle_tags),
      (gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result),
      (gst_rg_analysis_album_result):
      * gst/replaygain/gstrganalysis.h:
      * gst/replaygain/gstrglimiter.c: (gst_rg_limiter_base_init),
      (gst_rg_limiter_class_init), (gst_rg_limiter_init),
      (gst_rg_limiter_set_property), (gst_rg_limiter_get_property),
      (gst_rg_limiter_transform_ip):
      * gst/replaygain/gstrglimiter.h:
      * gst/replaygain/gstrgvolume.c: (gst_rg_volume_base_init),
      (gst_rg_volume_class_init), (gst_rg_volume_init),
      (gst_rg_volume_set_property), (gst_rg_volume_get_property),
      (gst_rg_volume_dispose), (gst_rg_volume_change_state),
      (gst_rg_volume_sink_event), (gst_rg_volume_tag_event),
      (gst_rg_volume_reset), (gst_rg_volume_update_gain),
      (gst_rg_volume_determine_gain):
      * gst/replaygain/gstrgvolume.h:
      * gst/replaygain/replaygain.c: (plugin_init):
      * gst/replaygain/replaygain.h:
      * gst/replaygain/rganalysis.h:
      * tests/check/Makefile.am:
      * tests/check/elements/.cvsignore:
      * tests/check/elements/rganalysis.c: (send_eos_event),
      (GST_START_TEST):
      * tests/check/elements/rglimiter.c: (setup_rglimiter),
      (cleanup_rglimiter), (set_playing_state), (create_test_buffer),
      (verify_test_buffer), (GST_START_TEST), (rglimiter_suite), (main):
      * tests/check/elements/rgvolume.c: (event_func), (setup_rgvolume),
      (cleanup_rgvolume), (set_playing_state), (set_null_state),
      (send_eos_event), (send_tag_event), (test_buffer_new),
      (fail_unless_target_gain), (fail_unless_result_gain),
      (fail_unless_gain), (GST_START_TEST), (rgvolume_suite), (main):
      Add replaygain playback elements (#412710).
      4e45e0a2
  4. 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
  5. 17 May, 2007 3 commits
    • 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
    • Edward Hervey's avatar
      sys/osxvideo/osxvideosink.*: Remove the event-loop-in-separate-thread... · 03897942
      Edward Hervey authored
      sys/osxvideo/osxvideosink.*: Remove the event-loop-in-separate-thread modifications, because MacOSX is $#@(*%$# ! For...
      
      Original commit message from CVS:
      * sys/osxvideo/osxvideosink.h:
      * sys/osxvideo/osxvideosink.m:
      Remove the event-loop-in-separate-thread modifications, because MacOSX
      is $#@(*%$# ! For those wondering, the event handling needs to be done
      in the main thread after all..
      03897942
    • Edward Hervey's avatar
      sys/osxvideo/osxvideosink.*: Fix a stupid #if vs #ifdef bug. Should use the proper colorspace now. · 284c7f0f
      Edward Hervey authored
      Original commit message from CVS:
      * sys/osxvideo/osxvideosink.h:
      * sys/osxvideo/osxvideosink.m:
      Fix a stupid #if vs #ifdef bug. Should use the proper colorspace now.
      Use a separate thread/task for the cocoa event_loop, else it wouldn't
      stop.
      284c7f0f
  6. 16 May, 2007 2 commits
  7. 15 May, 2007 5 commits
  8. 14 May, 2007 5 commits
    • Wim Taymans's avatar
      gst/rtp/: Update theora pay/depayloader in a similar to vorbis. · 4da361f9
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstrtptheoradepay.c: (decode_base64),
      (gst_rtp_theora_depay_parse_configuration):
      * gst/rtp/gstrtptheorapay.c: (encode_base64),
      (gst_rtp_theora_pay_finish_headers),
      (gst_rtp_theora_pay_handle_buffer):
      Update theora pay/depayloader in a similar to vorbis.
      * gst/rtp/gstrtpvorbisdepay.c:
      (gst_rtp_vorbis_depay_parse_configuration):
      Update docs.
      4da361f9
    • 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/rtp/gstrtpvorbisdepay.c: Remove annoying _dump_mem. · 4333477d
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
      Remove annoying _dump_mem.
      4333477d
    • 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
  9. 13 May, 2007 2 commits
  10. 12 May, 2007 3 commits
    • Peter Kjellerstedt's avatar
      gst/rtsp/: Make channel guint8 where possible. · 7ef62aac
      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/gstrtspsrc.h:
      * gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
      * gst/rtsp/rtspmessage.c: (rtsp_message_init_data),
      (rtsp_message_get_header):
      * gst/rtsp/rtspmessage.h:
      Make channel guint8 where possible.
      Make rtsp_message_init_data() take the channel as a guint8.
      * gst/rtsp/rtspdefs.c:
      Fixed a typo: Timout -> Timeout
      * gst/rtsp/rtspdefs.h:
      Make RTSP_CHECK() behave as a statement.
      * gst/rtsp/sdpmessage.c:
      Avoid a compiler warning in INIT_ARRAY().
      Fixes #437692.
      7ef62aac
    • Peter Kjellerstedt's avatar
      gst/rtsp/rtspurl.*: Add support for query parameters to RTSP URLs. · 02a64fe5
      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/rtspurl.c: (rtsp_url_parse), (rtsp_url_free),
      (rtsp_url_get_request_uri):
      * gst/rtsp/rtspurl.h:
      Add support for query parameters to RTSP URLs.
      02a64fe5
    • Peter Kjellerstedt's avatar
      gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse(). · 5f9984e8
      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/rtsptransport.c: (rtsp_transport_init), (parse_mode),
      (parse_range), (range_as_text), (rtsp_transport_mode_as_text),
      (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
      (rtsp_transport_parse), (rtsp_transport_as_text):
      * gst/rtsp/rtsptransport.h:
      Add validation to rtsp_transport_parse().
      Add rtsp_transport_as_text() to generate an RTSP header from an
      RTSPTransport.
      Change ssrc to guint (was a string) since that is what it is, even
      though it is sent as a hex string.
      Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
      incorrect, which can be seen when looking at the examples in the RFC).
      Fixes #437670.
      5f9984e8
  11. 11 May, 2007 7 commits
    • Eric Anholt's avatar
      sys/ximage/gstximagesrc.c (gst_ximage_src_open_display, gst_ximage_src_ximage_get): · 28713ecd
      Eric Anholt authored and Zaheer Abbas Merali's avatar Zaheer Abbas Merali committed
      Original commit message from CVS:
      Patch by: Eric Anholt
      * sys/ximage/gstximagesrc.c (gst_ximage_src_open_display,
      gst_ximage_src_ximage_get):
      Use union of all damage between frames to make it faster.
      Fixes bug #342463.
      Also fix crasher when cursor is at bottom right of window.
      28713ecd
    • Tim-Philipp Müller's avatar
      gst/wavparse/gstwavparse.c: Skip LIST chunks before the fmt chunk (fixes... · 4128e375
      Tim-Philipp Müller authored
      gst/wavparse/gstwavparse.c: Skip LIST chunks before the fmt chunk (fixes #437499). Also fix streaming mode regression...
      
      Original commit message from CVS:
      * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
      Skip LIST chunks before the fmt chunk (fixes #437499). Also fix
      streaming mode regression for file from #343837 with 'bext' chunk
      before the 'fmt' chunk.
      4128e375
    • 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
    • Wim Taymans's avatar
      gst/rtp/README: Update README with new RTP variables that will be used for synchronisation. · 5bc71b66
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtp/README:
      Update README with new RTP variables that will be used for
      synchronisation.
      * gst/rtp/gstrtpvorbisdepay.c: (decode_base64),
      (gst_rtp_vorbis_depay_parse_configuration),
      (gst_rtp_vorbis_depay_process):
      * gst/rtp/gstrtpvorbispay.c: (encode_base64),
      (gst_rtp_vorbis_pay_finish_headers),
      (gst_rtp_vorbis_pay_handle_buffer):
      Update vorbis pay and depayloader to draft-04.
      5bc71b66
    • Wim Taymans's avatar
      gst/rtsp/rtsptransport.c: UDP MCAST is actually the default for RTP/AVP. · 3e1fd612
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/rtsptransport.c:
      UDP MCAST is actually the default for RTP/AVP.
      3e1fd612
    • Zaheer Abbas Merali's avatar
      sys/ximage/gstximagesrc.c (gst_ximage_src_start, gst_ximage_src_ximage_get): · 20bc2905
      Zaheer Abbas Merali authored
      Original commit message from CVS:
      * sys/ximage/gstximagesrc.c (gst_ximage_src_start,
      gst_ximage_src_ximage_get):
      * sys/ximage/gstximagesrc.h (last_ximage):
      When using Damage actually keep the last frame, and not assume
      that the buffer we get already has the last frame on it.
      Copy the cursor over if we specify a non-zero start x and
      start y.
      20bc2905
    • Wim Taymans's avatar
      gst/rtsp/rtsptransport.c: Make UDP the default transport when not specified. · 4b69fc44
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtsp/rtsptransport.c:
      Make UDP the default transport when not specified.
      4b69fc44
  12. 10 May, 2007 1 commit
    • Stefan Kost's avatar
      gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header,... · eb5b5a84
      Stefan Kost authored
      gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde...
      
      Original commit message from CVS:
      * gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream,
      gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment,
      gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows,
      gst_qtdemux_loop_state_movie, gst_qtdemux_loop,
      qtdemux_parse_segments, qtdemux_parse_trak):
      * gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth,
      rtp_session_get_rtcp_bandwidth, rtp_session_get_cname,
      rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone,
      rtp_session_get_location, rtp_session_get_tool,
      rtp_session_process_bye, session_report_blocks):
      * gst/rtpmanager/rtpsource.c (rtp_source_process_rtp,
      rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb):
      More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>).
      * gst/switch/Makefile.am:
      Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).
      eb5b5a84