1. 05 Jun, 2007 1 commit
  2. 31 May, 2007 2 commits
  3. 29 May, 2007 1 commit
  4. 25 May, 2007 1 commit
    • Jan Schmidt's avatar
      Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release. · 588bc09c
      Jan Schmidt authored
      Original commit message from CVS:
      * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
      (gst_alsa_mixer_free), (gst_alsa_mixer_update),
      (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
      (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
      (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
      * ext/alsa/gstalsamixer.h:
      * ext/alsa/gstalsamixerelement.c:
      (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:
      Revert commits towards #152864 made so far. We'll pick it up again
      after the 0.10.13 release.
      588bc09c
  5. 24 May, 2007 4 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush)... · b2fdf703
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render):
      After an interrupt (PAUSED/flush) assume that the next sample should not
      be aligned to the previous sample. Fixes #417992.
      b2fdf703
    • Tim-Philipp Müller's avatar
      gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the... · 57375cf6
      Tim-Philipp Müller authored
      gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
      
      Original commit message from CVS:
      * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
      Don't add channels and rate fields to the template caps for
      audio/x-dts, as wavparse might not always be able to set them,
      which would then lead to 'caps are not a real subset of the
      template caps' warnings.
      57375cf6
    • Jan Schmidt's avatar
      gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without... · d9504cf0
      Jan Schmidt authored
      gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
      
      Original commit message from CVS:
      * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
      Handle unknown or invalid pads without crashing, as might occur if
      a media file like an mp3 is specified as a subtitle file.
      Fixes: #410039
      d9504cf0
    • Jan Schmidt's avatar
      gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting... · c446f911
      Jan Schmidt authored
      gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
      
      Original commit message from CVS:
      * gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
      (setup_sinks):
      Block the subtitle bin output queue before ghosting it and linking,
      then unblock after. This avoids spurious not-linked errors caused
      by the queue starting up (because it gets linked when it is ghosted).
      Fixes: #350299
      c446f911
  6. 23 May, 2007 1 commit
    • Jan Schmidt's avatar
      tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to... · e1cacbdc
      Jan Schmidt authored
      tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
      
      Original commit message from CVS:
      * tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
      Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
      file. Avoids flukes where the input gets typefound to some valid but
      useless type.
      e1cacbdc
  7. 22 May, 2007 4 commits
    • Tim-Philipp Müller's avatar
      tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs. · b0c7ebb4
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * tests/check/Makefile.am:
      * tests/check/elements/.cvsignore:
      * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
      (cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
      Add unit test for gnomevfssink seeking and position reporting for
      file:// URIs.
      b0c7ebb4
    • Mark Nauwelaerts's avatar
      ext/gnomevfs/gstgnomevfssink.*: see #412648. · b274e57b
      Mark Nauwelaerts authored
      Original commit message from CVS:
      Patch by: Mark Nauwelaerts <manauw at skynet be>
      * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
      (gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
      (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
      * ext/gnomevfs/gstgnomevfssink.h:
      Fix position reporting, especially after a seek (from upstream),
      see #412648.
      b274e57b
    • Tim-Philipp Müller's avatar
      ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut. · 1273d02f
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * ext/cdparanoia/gstcdparanoiasrc.c:
      Repair umlaut.
      1273d02f
    • Jan Schmidt's avatar
      gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3... · bec7949e
      Jan Schmidt authored
      gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
      
      Original commit message from CVS:
      * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
      Specify the full valid range for MP3 samplerates. Fixes a regression
      caused by extra header checks since the last release.
      bec7949e
  8. 21 May, 2007 4 commits
    • Mike Smith's avatar
      sys/: Fix a locking-order bug I introduced with my changes the other day. · cfc44030
      Mike Smith authored
      Original commit message from CVS:
      * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
      * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
      Fix a locking-order bug I introduced with my changes the other day.
      Patch by Mike Smith.
      cfc44030
    • Michael Smith's avatar
      ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames) · b48b9fdc
      Michael Smith authored
      Original commit message from CVS:
      * ext/theora/theoradec.c: (theora_handle_data_packet):
      Don't look inside 0-length packets (which indicate duplicated
      frames)
      b48b9fdc
    • Wim Taymans's avatar
      Small cleanups. · 9b188adc
      Wim Taymans authored
      Original commit message from CVS:
      * ext/cdparanoia/gstcdparanoiasrc.c:
      (gst_cd_paranoia_src_read_sector):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create):
      Small cleanups.
      * ext/theora/theoradec.c: (theora_dec_sink_event):
      Fix typo.
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_set_gst_timestamp):
      Add some FIXME
      * gst/playback/gstdecodebin.c: (queue_underrun_cb):
      And some debug info when a FIXME path is hit.
      9b188adc
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime... · 7ace8599
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
      (gst_base_rtp_audio_payload_class_init),
      (gst_base_rtp_audio_payload_init),
      (gst_base_rtp_audio_payload_finalize),
      (gst_base_rtp_audio_payload_handle_frame_based_buffer),
      (gst_base_rtp_audio_payload_handle_sample_based_buffer),
      (gst_base_rtp_payload_audio_handle_event):
      Some cleanups, remove minptime property as it is now in the parent
      class.
      Override parent class event function.
      * gst-libs/gst/rtp/gstbasertppayload.c:
      (gst_basertppayload_class_init), (gst_basertppayload_init),
      (gst_basertppayload_event), (gst_basertppayload_set_property),
      (gst_basertppayload_get_property):
      * gst-libs/gst/rtp/gstbasertppayload.h:
      Add min-ptime property.
      Add handle-event vmethod. Fixes #415001.
      7ace8599
  9. 18 May, 2007 3 commits
    • Christian Schaller's avatar
      update spec · 13b6bdee
      Christian Schaller authored
      Original commit message from CVS:
      update spec
      13b6bdee
    • Stefan Kost's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c · e7c3ddf3
      Stefan Kost authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c
      (gst_base_audio_sink_change_state):
      Fix typo in comment.
      * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
      free_dynamics, pad_probe, close_pad_link, try_to_link_1,
      get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
      close_link):
      * gst/playback/gstplaybin.c (gst_play_bin_set_property,
      gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
      Remove trailing whitespaces in comments.
      * gst/volume/Makefile.am:
      Fix tabs.
      e7c3ddf3
    • Marc-Andre Lureau's avatar
      gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option,... · 16b8bd4c
      Marc-Andre Lureau authored
      gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
      
      Original commit message from CVS:
      patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
      * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
      set_option, get_option, _gst_reserved):
      Revert reordering functions (keep ABI).
      16b8bd4c
  10. 17 May, 2007 8 commits
    • Jan Schmidt's avatar
      sys/: When we create our own window, indicate that we handle the · cbc95dfb
      Jan Schmidt authored
      Original commit message from CVS:
      * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
      (gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
      (gst_ximagesink_show_frame):
      * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
      (gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
      (gst_xvimagesink_show_frame):
      When we create our own window, indicate that we handle the
      WM_DELETE client message from the window manager, so that it won't
      kill our window (and our app) along with it. Handle ClientMessage,
      post an error on the bus, and close the window. Further buffers
      arriving will result in a FlowError because the window has been
      destroyed.
      Fixes: #393975
      Clean up the X event handling loop and make them the same for
      both xvimagesink and ximagesink while I'm at it.
      cbc95dfb
    • Wim Taymans's avatar
      gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too. · a18a10e8
      Wim Taymans authored
      Original commit message from CVS:
      * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
      Make decodebin2 autoplug depayloaders too.
      * gst/playback/gsturidecodebin.c: (source_new_pad):
      Set the newly created decoder in a usable state when autoplugging a
      dynamic source such as RTSP.
      a18a10e8
    • Tim-Philipp Müller's avatar
      gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and... · 2cd5f527
      Tim-Philipp Müller authored
      gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
      
      Original commit message from CVS:
      * gst/playback/gststreaminfo.c: (cb_probe):
      Ignore video-codec tag for audio streams and ignore audio-codec tags
      for video streams. Should make codec name collection a bit more
      robust against sloppy demuxers that send tag events containing both
      tags down each pad.
      2cd5f527
    • Wim Taymans's avatar
      gst/playback/gstqueue2.c: Tweak the buffering thresholds a little. · d3393980
      Wim Taymans authored
      Original commit message from CVS:
      * gst/playback/gstqueue2.c: (update_rates):
      Tweak the buffering thresholds a little.
      Update the buffer size with the previously calculate rate instead of
      only when we calculate a new rate so that we get smoother buffering
      updates.
      * gst/playback/Makefile.am:
      * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
      (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
      (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
      (gst_uri_decode_bin_get_property), (unknown_type),
      (add_element_stream), (no_more_pads_full), (no_more_pads),
      (source_no_more_pads), (new_decoded_pad), (array_has_value),
      (gen_source_element), (has_all_raw_caps), (analyse_source),
      (remove_decoders), (make_decoder), (remove_source),
      (source_new_pad), (setup_source), (decoder_query_init),
      (decoder_query_duration_fold), (decoder_query_duration_done),
      (decoder_query_position_fold), (decoder_query_position_done),
      (decoder_query_latency_fold), (decoder_query_latency_done),
      (decoder_query_seeking_fold), (decoder_query_seeking_done),
      (decoder_query_generic_fold), (gst_uri_decode_bin_query),
      (gst_uri_decode_bin_change_state), (plugin_init):
      New element that intergrates a source, optional buffering element and
      decodebin.
      d3393980
    • Tim-Philipp Müller's avatar
      configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check... · 23396338
      Tim-Philipp Müller authored
      configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
      
      Original commit message from CVS:
      * configure.ac:
      Bump libtheora requirement to 1.0alpha5 for the pixformat check
      (also has a .pc file, so we don't need the fallback check any
      longer). Fixes #438840.
      23396338
    • Wim Taymans's avatar
      gst/playback/gstqueue2.c: fix build. · fa972968
      Wim Taymans authored
      Original commit message from CVS:
      * gst/playback/gstqueue2.c: (gst_queue_get_type),
      (gst_queue_class_init), (gst_queue_finalize), (update_time_level),
      (apply_segment), (apply_buffer), (update_buffering),
      (reset_rate_timer), (update_rates), (gst_queue_locked_flush),
      (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
      (gst_queue_handle_sink_event), (gst_queue_is_filled),
      (gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
      (plugin_init):
      fix build.
      fa972968
    • Wim Taymans's avatar
      gst/playback/: On our way to playbin2 this is the new network queue that does... · ae69903c
      Wim Taymans authored
      gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
      
      Original commit message from CVS:
      * gst/playback/Makefile.am:
      * gst/playback/gstqueue2.c: (gst_queue_get_type),
      (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
      (gst_queue_getcaps), (gst_queue_bufferalloc),
      (gst_queue_acceptcaps), (update_time_level), (apply_segment),
      (apply_buffer), (update_buffering), (reset_rate_timer),
      (update_rates), (gst_queue_locked_flush),
      (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
      (gst_queue_handle_sink_event), (gst_queue_is_empty),
      (gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
      (gst_queue_loop), (gst_queue_handle_src_event),
      (gst_queue_handle_src_query), (gst_queue_sink_activate_push),
      (gst_queue_src_activate_push), (gst_queue_change_state),
      (gst_queue_set_property), (gst_queue_get_property), (plugin_init):
      On our way to playbin2 this is the new network queue that does buffering
      all by itself using high and low watermarks. It can also measure up and
      downstream bandwidth to optimally size the queue.
      ae69903c
    • 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
  11. 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
  12. 13 May, 2007 1 commit
  13. 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
  14. 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