1. 17 May, 2007 1 commit
    • Michael Smith's avatar
      gst/: Use the segment->last_stop value to calculate the next timestamp to... · ab76fa09
      Michael Smith authored
      gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
      
      Original commit message from CVS:
      * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
      * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
      Use the segment->last_stop value to calculate the next timestamp to
      generate after a seek; not the segment->start value.
      ab76fa09
  2. 15 May, 2007 7 commits
    • David Schleef's avatar
      docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This... · bd9d834b
      David Schleef authored
      docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled.  This matches the behavior of gtk+.  Fixes #3...
      
      Original commit message from CVS:
      * docs/Makefile.am: Install docs even when --disable-gtk-doc
      is disabled.  This matches the behavior of gtk+.  Fixes #349099.
      bd9d834b
    • Wim Taymans's avatar
      ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes. · f8f9935d
      Wim Taymans authored
      Original commit message from CVS:
      * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
      (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
      Some more chained streaming ogg timestamp fixes.
      f8f9935d
    • Wim Taymans's avatar
      ext/ogg/gstoggdemux.c: Add some FIXMEs. · 8b90454e
      Wim Taymans authored
      Original commit message from CVS:
      * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
      (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
      (gst_ogg_demux_handle_page):
      Add some FIXMEs.
      Fix chain start/stop segment handling based on patch by
      <ahalda at cs dot mcgill dot ca> see #320984.
      8b90454e
    • Michael Smith's avatar
      configure.ac: We don't require a C++ compiler. So don't require one. · 171fb33d
      Michael Smith authored
      Original commit message from CVS:
      * configure.ac:
      We don't require a C++ compiler. So don't require one.
      171fb33d
    • Stefan Kost's avatar
      ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check,... · 38da6419
      Stefan Kost authored
      ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
      
      Original commit message from CVS:
      * ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
      gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
      gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
      gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
      gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
      gst_alsa_mixer_update_track):
      Apply some of the cleanup Tim suggested in #152864 afterwards.
      38da6419
    • Marc-Andre Lureau's avatar
      ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch,... · f2df2a69
      Marc-Andre Lureau authored
      ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
      
      Original commit message from CVS:
      patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
      * ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
      _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
      gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
      gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
      gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
      gst_alsa_mixer_handle_source_callback,
      gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
      gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
      gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
      gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
      gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
      gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
      * ext/alsa/gstalsamixer.h (handle_source, interface, dir):
      * ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
      gst_alsa_mixer_element_interface_supported,
      gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
      gst_alsa_mixer_element_set_property,
      gst_alsa_mixer_element_get_property,
      gst_alsa_mixer_element_change_state):
      * ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
      * gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
      gst_mixer_option_changed):
      * gst-libs/gst/interfaces/mixer.h (set_option, get_option,
      volume_changed, option_changed, _gst_reserved):
      Implement notification for alsamixer. Fixes #152864
      f2df2a69
    • David Schleef's avatar
      gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer. · c655a27a
      David Schleef authored
      Original commit message from CVS:
      * gst/videotestsrc/videotestsrc.c:
      * gst/videotestsrc/videotestsrc.h:
      Add support for video/x-raw-bayer.
      c655a27a
  3. 13 May, 2007 1 commit
  4. 12 May, 2007 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields... · 01b6f0b3
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_setcaps),
      (gst_base_rtp_depayload_set_gst_timestamp):
      Parse and use additional caps fields as described in updated
      application/x-rtp caps spec.
      01b6f0b3
    • Wim Taymans's avatar
      ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data... · 8532e91e
      Wim Taymans authored
      ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
      
      Original commit message from CVS:
      * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
      (gst_ogg_demux_collect_chain_info):
      If there is a stream in a chain without any data packets, ignore the
      stream in the total length calculations. Might be related to #436820.
      8532e91e
  5. 11 May, 2007 1 commit
    • Jan Schmidt's avatar
      gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system... · 1e2c3277
      Jan Schmidt authored
      gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
      
      Original commit message from CVS:
      * gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
      (mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
      (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
      (mpeg_video_type_find), (mpeg_video_stream_type_find),
      (plugin_init):
      Consolidate and re-work our mpeg system stream detection to probe
      more packets and produce a higher confidence result. Fixes a
      regression caused by lowering the typefind probability last year
      - related to bug #397810. Remove the redundant MPEG-1 specific
      typefind function, as the new one detects both MPEG-1 & MPEG-2
      happily.
      Also cleanup the MPEG elementary and MPEG-TS detection functions a
      little.
      Tested against my media test directory, with some improvements and
      no regressions.
      1e2c3277
  6. 10 May, 2007 1 commit
    • Wim Taymans's avatar
      gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal... · 56f01bc0
      Wim Taymans authored
      gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
      
      Original commit message from CVS:
      * gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
      (queue_out_of_data):
      Connect to the new queue "pushing" signal instead of the broken
      "running" one.
      56f01bc0
  7. 09 May, 2007 3 commits
  8. 08 May, 2007 1 commit
    • Stefan Kost's avatar
      gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,... · 736a5c08
      Stefan Kost authored
      gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
      
      Original commit message from CVS:
      * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
      gst_adder_change_state):
      * gst/adder/gstadder.h (bps, offset, collect_event, segment,
      segment_pending, segment_position, segment_rate):
      Handle playback-rate on adder.
      736a5c08
  9. 07 May, 2007 1 commit
    • Michael Smith's avatar
      ext/theora/: Don't push events (newsegment, tags) before initialising the decoder. · db624feb
      Michael Smith authored
      Original commit message from CVS:
      * ext/theora/gsttheoradec.h:
      * ext/theora/theoradec.c: (gst_theora_dec_reset),
      (theora_dec_sink_event), (theora_handle_comment_packet),
      (theora_handle_type_packet), (theora_dec_change_state):
      Don't push events (newsegment, tags) before initialising the
      decoder.
      This is neccesary for seeking to work correctly in gnonlin.
      db624feb
  10. 04 May, 2007 4 commits
  11. 03 May, 2007 4 commits
    • Tim-Philipp Müller's avatar
      sys/ximage/ximagesink.c: When XShm is not available, we might get row strides... · cb73a6e7
      Tim-Philipp Müller authored
      sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
      
      Original commit message from CVS:
      * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
      When XShm is not available, we might get row strides that are not
      rounded up to multiples of four; this is bad, because virtually
      every RGB-processing element in GStreamer assumes rowstrides are
      rounded up to multiples of four, so let's allocate at least enough
      memory to avoid crashes in this case. The image will still be
      displayed distorted though if this happens, so that still needs
      fixing (maybe by allocating a bigger image with an 'even' width
      and then clipping it appropriately when rendering - something for
      Xlib aficionados in any case).
      cb73a6e7
    • Michael Smith's avatar
      gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume... · 03e4592e
      Michael Smith authored
      gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
      
      Original commit message from CVS:
      * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
      If a buffer doesn't have a timestamp, assume it's contiguous with
      the previous buffer, and synthesise timestamps appropriately.
      03e4592e
    • Edward Hervey's avatar
      tests/check/elements/videorate.c: Set buffer timestamp to a valid value in... · 14f2bca5
      Edward Hervey authored
      tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
      
      Original commit message from CVS:
      * tests/check/elements/videorate.c: (GST_START_TEST):
      Set buffer timestamp to a valid value in order to test the buffer
      really does stay in videorate.
      14f2bca5
    • Edward Hervey's avatar
      gst/videorate/gstvideorate.c: There is no sensible way to handle incoming... · 25d28aae
      Edward Hervey authored
      gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
      
      Original commit message from CVS:
      * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
      There is no sensible way to handle incoming buffers which don't have a
      valid timestamp. We therefore discard them and wait for the next one.
      25d28aae
  12. 01 May, 2007 1 commit
  13. 29 Apr, 2007 1 commit
  14. 27 Apr, 2007 2 commits
    • Julien Moutte's avatar
      ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888). · d299d1c0
      Julien Moutte authored
      Original commit message from CVS:
      2007-04-27  Julien MOUTTE  <julien@moutte.net>
      
      * ext/theora/theoradec.c: (_theora_granule_time),
      (theora_dec_push_forward), (theora_handle_data_packet),
      (theora_dec_decode_buffer): Calculate buffer duration correctly
      to generate a perfect stream (#433888).
      * gst/audioresample/gstaudioresample.c:
      (audioresample_check_discont): Glib provides ABS.
      d299d1c0
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing. · f23356bd
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
      (gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
      (gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
      (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
      (gst_rtcp_packet_bye_set_reason):
      * gst-libs/gst/rtp/gstrtcpbuffer.h:
      Fix RB block parsing and writing.
      Add support for constructing BYE packets.
      f23356bd
  15. 25 Apr, 2007 2 commits
    • Tim-Philipp Müller's avatar
      When posting a warning message because samples were dropped, post something... · 9e873a3c
      Tim-Philipp Müller authored
      When posting a warning message because samples were dropped, post something more intelligible than he default error m...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
      (gst_base_audio_src_create):
      * po/POTFILES.in:
      When posting a warning message because samples were dropped, post
      something more intelligible than he default error message for clock
      errors which is just confusing in this context (#432984).
      9e873a3c
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets. · f5c743b0
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
      (gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
      (read_packet_header), (gst_rtcp_packet_move_to_next),
      (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
      (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
      (gst_rtcp_packet_sdes_get_item_count),
      (gst_rtcp_packet_sdes_first_item),
      (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
      (gst_rtcp_packet_sdes_first_entry),
      (gst_rtcp_packet_sdes_next_entry),
      (gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
      (gst_rtcp_packet_sdes_add_entry):
      * gst-libs/gst/rtp/gstrtcpbuffer.h:
      Implement code to write SR, RR and SDES packets.
      f5c743b0
  16. 24 Apr, 2007 3 commits
    • Christian Kirbach's avatar
      sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362). · 80b16b3a
      Christian Kirbach authored
      Original commit message from CVS:
      Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
      * sys/ximage/ximagesink.c:
      Fix build if XShm is not available (#432362).
      80b16b3a
    • Sebastian Dröge's avatar
      gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes,... · 84c824b9
      Sebastian Dröge authored
      gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
      
      Original commit message from CVS:
      * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
      Initalize the AudioConvertCtx with zeroes, otherwise it will contain
      pointers to random memory which are passed to g_free() when
      audio_convert_prepare_context() is called the first time.
      84c824b9
    • Dan Williams's avatar
      gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push()... · 37a334dd
      Dan Williams authored
      gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
      
      Original commit message from CVS:
      Patch by: Dan Williams <dcbw redhat com>
      * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
      Don't leak incoming buffer if gst_pad_push() returns a
      non-OK flow. Fixes #432755.
      * tests/check/elements/videorate.c: (GST_START_TEST),
      (videorate_suite):
      Unit test for the above by Yours Truly.
      37a334dd
  17. 23 Apr, 2007 1 commit
  18. 21 Apr, 2007 4 commits
    • Tim-Philipp Müller's avatar
      ChangeLog surgery: add API keyword · 478fd777
      Tim-Philipp Müller authored
      Original commit message from CVS:
      ChangeLog surgery: add API keyword
      478fd777
    • Olivier Crete's avatar
      gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose... · e3ff444d
      Olivier Crete authored
      gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
      
      Original commit message from CVS:
      Patch by: Olivier Crete  <tester at tester ca>
      * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
      (gst_base_rtp_audio_payload_class_init),
      (gst_base_rtp_audio_payload_init),
      (gst_base_rtp_audio_payload_dispose):
      Chain up to parent class in dispose function; get rid of
      unnecessary 'diposed' flag in private structure (#415001).
      e3ff444d
    • Tim-Philipp Müller's avatar
      Some minor docs fixes and additions; also add missing 'Since' bits. · 71d77fbe
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs.types:
      * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
      (gst_base_rtp_audio_payload_class_init):
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      * gst-libs/gst/rtp/gstbasertppayload.c:
      Some minor docs fixes and additions; also add missing 'Since' bits.
      71d77fbe
    • Zeeshan Ali's avatar
      gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added... · 80ebb9eb
      Zeeshan Ali authored
      gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
      
      Original commit message from CVS:
      Patch by: Zeeshan Ali  <zeenix gmail com>
      * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
      (gst_base_rtp_audio_payload_handle_frame_based_buffer),
      (gst_base_rtp_audio_payload_handle_sample_based_buffer),
      (gst_base_rtp_audio_payload_push):
      * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
      The recently-added gst_base_rtp_audio_payload_push() should take an
      object of type GstBaseRTPAudioPayload as first argument (#431672).
      80ebb9eb