1. 28 Aug, 2005 1 commit
  2. 26 Aug, 2005 8 commits
    • Wim Taymans's avatar
      gst/audioconvert/audioconvert.c: Cleanups. · b6c368ce
      Wim Taymans authored
      Original commit message from CVS:
      * gst/audioconvert/audioconvert.c: (if), (float),
      (audio_convert_get_func_index), (check_default),
      (audio_convert_clean_fmt), (audio_convert_prepare_context),
      (audio_convert_clean_context), (audio_convert_get_sizes),
      (audio_convert_convert):
      Cleanups.
      b6c368ce
    • Wim Taymans's avatar
      gst/audioconvert/audioconvert.c: More elegant and working temp buffer selection algo. · ddec57c0
      Wim Taymans authored
      Original commit message from CVS:
      * gst/audioconvert/audioconvert.c: (if), (float),
      (audio_convert_get_func_index), (check_default),
      (audio_convert_clean_fmt), (audio_convert_prepare_context),
      (audio_convert_clean_context), (audio_convert_get_sizes),
      (audio_convert_convert):
      More elegant and working temp buffer selection algo.
      ddec57c0
    • Wim Taymans's avatar
      gst/audioconvert/audioconvert.c: Use realloc else we lose our original data. · 123aa7de
      Wim Taymans authored
      Original commit message from CVS:
      * gst/audioconvert/audioconvert.c: (if), (float),
      (audio_convert_get_func_index), (check_default),
      (audio_convert_clean_fmt), (audio_convert_prepare_context),
      (audio_convert_clean_context), (audio_convert_get_sizes),
      (get_temp_buffer), (audio_convert_convert):
      Use realloc else we lose our original data.
      123aa7de
    • Thomas Vander Stichele's avatar
      use base class' newsegment to properly timestamp · f0f2b133
      Thomas Vander Stichele authored
      Original commit message from CVS:
      
      use base class' newsegment to properly timestamp
      f0f2b133
    • Wim Taymans's avatar
      gst/audioconvert/: Oops, allocate enough space to perform the channel mix. · 98fbd82d
      Wim Taymans authored
      Original commit message from CVS:
      * gst/audioconvert/audioconvert.c: (if), (float),
      (audio_convert_get_func_index), (check_default),
      (audio_convert_clean_fmt), (audio_convert_prepare_context),
      (audio_convert_clean_context), (audio_convert_get_sizes),
      (get_temp_buffer), (audio_convert_convert):
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_parse_caps), (gst_audio_convert_get_unit_size),
      (gst_audio_convert_transform_caps),
      (gst_audio_convert_fixate_caps), (gst_audio_convert_transform):
      * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_mix):
      Oops, allocate enough space to perform the channel mix.
      98fbd82d
    • Wim Taymans's avatar
      gst/audioconvert/: Cleanups, librarify a bit, optimize, better negotiation and more. · ceb84de9
      Wim Taymans authored
      Original commit message from CVS:
      * gst/audioconvert/Makefile.am:
      * gst/audioconvert/audioconvert.c: (if), (float),
      (audio_convert_get_func_index), (check_default),
      (audio_convert_clean_fmt), (audio_convert_prepare_context),
      (audio_convert_clean_context), (audio_convert_get_sizes),
      (get_temp_buffer), (audio_convert_convert):
      * gst/audioconvert/audioconvert.h:
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_class_init), (gst_audio_convert_init),
      (gst_audio_convert_dispose), (gst_audio_convert_parse_caps),
      (gst_audio_convert_get_unit_size),
      (gst_audio_convert_transform_caps),
      (gst_audio_convert_fixate_caps), (gst_audio_convert_set_caps),
      (gst_audio_convert_transform_ip), (gst_audio_convert_transform):
      * gst/audioconvert/gstaudioconvert.h:
      * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
      (gst_channel_mix_fill_identical),
      (gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
      (gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
      (gst_channel_mix_fill_normalize), (gst_channel_mix_fill_matrix),
      (gst_channel_mix_setup_matrix), (gst_channel_mix_passthrough),
      (gst_channel_mix_mix):
      * gst/audioconvert/gstchannelmix.h:
      Cleanups, librarify a bit, optimize, better negotiation and more.
      ceb84de9
    • Jan Schmidt's avatar
      ext/ogg/gstoggdemux.c: Another from MikeS: · ee2bc937
      Jan Schmidt authored
      Original commit message from CVS:
      * ext/ogg/gstoggdemux.c: (ogg_find_peek):
      Another from MikeS:
      During typefinding, don't support negative offsets
      (offsets from the end of the stream) in our typefind->peek() function
      - nothing embedded in ogg ever needs them. However, we need to recognise
      those requests and reject them, otherwise we return invalid pointers.
      ee2bc937
    • Jan Schmidt's avatar
      ext/: Big shout-out to MikeS for fixing this giant memory leak. · 538eabd5
      Jan Schmidt authored
      Original commit message from CVS:
      * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
      * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
      (vorbisdec_finalize), (vorbis_handle_type_packet):
      Big shout-out to MikeS for fixing this giant memory leak.
      Huzzah!
      538eabd5
  3. 25 Aug, 2005 9 commits
    • Thomas Vander Stichele's avatar
      add more conversion tests · 3f478d73
      Thomas Vander Stichele authored
      Original commit message from CVS:
      add more conversion tests
      3f478d73
    • Thomas Vander Stichele's avatar
      add more tests · 2042b4f2
      Thomas Vander Stichele authored
      Original commit message from CVS:
      add more tests
      2042b4f2
    • Thomas Vander Stichele's avatar
      plug some leaks · 43332aed
      Thomas Vander Stichele authored
      Original commit message from CVS:
      plug some leaks
      43332aed
    • Thomas Vander Stichele's avatar
      check/: add a test for audioconvert · 6dff9c2c
      Thomas Vander Stichele authored
      Original commit message from CVS:
      
      * check/Makefile.am:
      * check/elements/audioconvert.c: (setup_audioconvert),
      (cleanup_audioconvert), (get_int_caps), (verify_convert),
      (GST_START_TEST), (audioconvert_suite), (main):
      add a test for audioconvert
      * gst/audioresample/gstaudioresample.c:
      * gst/audioresample/gstaudioresample.h:
      set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
      note that for buffers of 1/3 sec this means DURATION(c) is
      one nanosecond more than for a and b
      6dff9c2c
    • Thomas Vander Stichele's avatar
      some more testing for perfect streams · 8f3a11d6
      Thomas Vander Stichele authored
      Original commit message from CVS:
      some more testing for perfect streams
      8f3a11d6
    • Thomas Vander Stichele's avatar
      add a check for audioresample · eae12502
      Thomas Vander Stichele authored
      Original commit message from CVS:
      add a check for audioresample
      eae12502
    • Thomas Vander Stichele's avatar
      show some info on what's left in the queue · f7cb2ba6
      Thomas Vander Stichele authored
      Original commit message from CVS:
      show some info on what's left in the queue
      f7cb2ba6
    • Thomas Vander Stichele's avatar
      gst/audioresample/: add room for extra overlap samples when asked to transform... · 7647f7fc
      Thomas Vander Stichele authored
      gst/audioresample/: add room for extra overlap samples when asked to transform size protect against possible mem corr...
      
      Original commit message from CVS:
      * gst/audioresample/debug.c:
      * gst/audioresample/gstaudioresample.c:
      add room for extra overlap samples when asked to transform size
      protect against possible mem corruption and check for discrepancies
      between written size and outbuffer's size so we can warn for
      potential problems
      * gst/audioresample/resample.c: (resample_init),
      (resample_get_output_size_for_input), (resample_get_output_size),
      (resample_set_n_channels), (resample_set_format):
      set debug level based on RESAMPLE_DEBUG env var
      make sure that get_output_size* returns a whole number of
      sample_size
      set sample_size each time either channel or format is set
      * gst/audioresample/resample_chunk.c: (resample_scale_chunk):
      * gst/audioresample/resample_functable.c:
      (resample_scale_functable):
      * gst/audioresample/resample_ref.c: (resample_scale_ref):
      remove r->sample_size, it's done in resample.c now
      add some debugging to the ref implementation
      make sure we only give back bytes that are wholes of the sample
      size
      7647f7fc
    • Jan Schmidt's avatar
      gst/playback/gstplaybasebin.c: Revert unpopular change for GST_MESSAGE_SRC to GObject. · 2a13ddfd
      Jan Schmidt authored
      Original commit message from CVS:
      * gst/playback/gstplaybasebin.c: (fill_buffer):
      Revert unpopular change for GST_MESSAGE_SRC to GObject.
      2a13ddfd
  4. 24 Aug, 2005 18 commits
    • Stefan Kost's avatar
      gst/volume/gstvolume.c: made set_caps function static · be10c8f8
      Stefan Kost authored
      Original commit message from CVS:
      * gst/volume/gstvolume.c:
      made set_caps function static
      be10c8f8
    • Wim Taymans's avatar
      ext/vorbis/vorbisenc.c: Stop leaking taglists. · 963941df
      Wim Taymans authored
      Original commit message from CVS:
      * ext/vorbis/vorbisenc.c: (gst_vorbisenc_init),
      (gst_vorbisenc_change_state):
      Stop leaking taglists.
      963941df
    • Thomas Vander Stichele's avatar
      debugging fixes · 46e443bd
      Thomas Vander Stichele authored
      Original commit message from CVS:
      debugging fixes
      46e443bd
    • Thomas Vander Stichele's avatar
      translate me baby · ffc57169
      Thomas Vander Stichele authored
      Original commit message from CVS:
      translate me baby
      ffc57169
    • Wim Taymans's avatar
      ext/ogg/gstoggdemux.c: Parse seeking events better. · 7824216c
      Wim Taymans authored
      Original commit message from CVS:
      * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
      (gst_ogg_pad_event), (gst_ogg_demux_factory_filter),
      (gst_ogg_pad_submit_packet), (gst_ogg_chain_new),
      (gst_ogg_demux_init), (gst_ogg_demux_perform_seek),
      (gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
      (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
      Parse seeking events better.
      Unref static caps.
      Generate correct newsegment events, fixes seeking in live oggs.
      
      * ext/theora/theoradec.c: (theora_dec_src_query),
      (theora_dec_src_event), (theora_dec_src_getcaps),
      (theora_dec_sink_event), (theora_dec_push), (theora_dec_chain):
      Use newsegment values to report correct play time.
      
      * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
      (vorbis_dec_src_event), (vorbis_dec_sink_event):
      * ext/vorbis/vorbisdec.h:
      Parse and use newsegment values to report correct play time.
      
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_event), (gst_base_audio_sink_render):
      Clear ringbuffer on flush.
      Use newsegment values to calculate playback time.
      
      * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
      * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
      Basesink does newsegment calculations for us now.
      7824216c
    • Thomas Vander Stichele's avatar
      c/: add core's plugins to the mix so that playbin works · 2136419a
      Thomas Vander Stichele authored
      Original commit message from CVS:
      
      * check/Makefile.am:
      * configure.ac:
      add core's plugins to the mix so that playbin works
      * check/generic/states.c: (GST_START_TEST):
      set a 0 timeout on pipelines, so they don't force the next
      state change
      * gst/playback/gstplaybasebin.c: (setup_source), (prepare_output),
      (gst_play_base_bin_change_state):
      remove the crappy error handling and do GST error handling
      2136419a
    • Christian Schaller's avatar
      add audioresample to spec file · e520824f
      Christian Schaller authored
      Original commit message from CVS:
      add audioresample to spec file
      e520824f
    • Christian Schaller's avatar
      fix broken header setup in Makefile.am · eeffbe7a
      Christian Schaller authored
      Original commit message from CVS:
      fix broken header setup in Makefile.am
      eeffbe7a
    • Thomas Vander Stichele's avatar
      dist more · ebdf1ac2
      Thomas Vander Stichele authored
      Original commit message from CVS:
      dist more
      ebdf1ac2
    • Thomas Vander Stichele's avatar
      check/: add same test as to core, it bitches out on playbin atm. · 886b4367
      Thomas Vander Stichele authored
      Original commit message from CVS:
      * check/Makefile.am:
      * check/generic/states.c: (GST_START_TEST), (states_suite), (main):
      add same test as to core, it bitches out on playbin atm.
      886b4367
    • Wim Taymans's avatar
      configure.ac: Remove audioscale. · f3ef56e8
      Wim Taymans authored
      Original commit message from CVS:
      * configure.ac:
      Remove audioscale.
      f3ef56e8
    • Wim Taymans's avatar
      gst/videoscale/gstvideoscale.*: Refactor, make use of BaseTranform really well. · da25385e
      Wim Taymans authored
      Original commit message from CVS:
      * gst/videoscale/gstvideoscale.c: (gst_videoscale_init),
      (gst_videoscale_prepare_size), (parse_caps),
      (gst_videoscale_set_caps), (gst_videoscale_get_size),
      (gst_videoscale_prepare_image), (gst_videoscale_transform_ip),
      (gst_videoscale_transform):
      * gst/videoscale/gstvideoscale.h:
      Refactor, make use of BaseTranform really well.
      da25385e
    • Thomas Vander Stichele's avatar
      port audioresample to basetransform · 752a5919
      Thomas Vander Stichele authored
      Original commit message from CVS:
      port audioresample to basetransform
      752a5919
    • Thomas Vander Stichele's avatar
      port audioconvert to basetransform fix ffmpegcsp and videoscale for basetransform changes · 41a43b86
      Thomas Vander Stichele authored
      Original commit message from CVS:
      port audioconvert to basetransform
      fix ffmpegcsp and videoscale for basetransform changes
      41a43b86
    • Jan Schmidt's avatar
      check/Makefile.am: Add CHECK_CFLAGS and LDFLAGS · 80ad4cff
      Jan Schmidt authored
      Original commit message from CVS:
      * check/Makefile.am:
      Add CHECK_CFLAGS and LDFLAGS
      
      * gst/playback/gstplaybasebin.c: (fill_buffer):
      GST_MESSAGE_SRC became a GObject
      80ad4cff
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstringbuffer.*: Added function to clear the ringbuffer. · 5ac2327f
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_set_sample),
      (gst_ring_buffer_clear_all):
      * gst-libs/gst/audio/gstringbuffer.h:
      Added function to clear the ringbuffer.
      5ac2327f
    • Andy Wingo Wingo's avatar
      whoops · b45323de
      Andy Wingo Wingo authored
      Original commit message from CVS:
      whoops
      b45323de
    • Andy Wingo Wingo's avatar
      sys/v4l/gstv4lelement.c (gst_v4lelement_start) · 7b9a366d
      Andy Wingo Wingo authored
      Original commit message from CVS:
      2005-08-24  Andy Wingo  <wingo@pobox.com>
      
      * sys/v4l/gstv4lelement.c (gst_v4lelement_start)
      (gst_v4lelement_stop): Call _start and _stop for xoverlay instead
      of _open and _close.
      
      * sys/v4l/gstv4lxoverlay.h:
      * sys/v4l/gstv4lxoverlay.c (gst_v4l_xoverlay_set_xwindow_id): Open
      an Xv connection here, instead of all the time. Make Xv only be
      loaded if you axe for it. Kindof a workaround for buggy behaviour
      of Xv when using remote xservers (XvQueryExtension would block).
      (gst_v4l_xoverlay_stop, gst_v4l_xoverlay_start): New functions,
      replace the _open and _close public API. Only start the xv
      connection if necessary.
      (gst_v4l_xoverlay_open, gst_v4l_xoverlay_close): Made static.
      7b9a366d
  5. 23 Aug, 2005 4 commits
    • David Schleef's avatar
      gst/audioresample/Makefile.am: Leet audioresampling code · ae8f41b6
      David Schleef authored
      Original commit message from CVS:
      * gst/audioresample/Makefile.am: Leet audioresampling code
      * gst/audioresample/buffer.c:
      * gst/audioresample/buffer.h:
      * gst/audioresample/debug.c:
      * gst/audioresample/debug.h:
      * gst/audioresample/functable.c:
      * gst/audioresample/functable.h:
      * gst/audioresample/gstaudioresample.c:
      * gst/audioresample/gstaudioresample.h:
      * gst/audioresample/resample.c:
      * gst/audioresample/resample.h:
      * gst/audioresample/resample_chunk.c:
      * gst/audioresample/resample_functable.c:
      * gst/audioresample/resample_ref.c:
      ae8f41b6
    • Wim Taymans's avatar
      examples/seeking/seek.c: Small seek updates. · 84d0eb4f
      Wim Taymans authored
      Original commit message from CVS:
      * examples/seeking/seek.c: (make_vorbis_pipeline),
      (make_theora_pipeline), (make_vorbis_theora_pipeline), (do_seek):
      Small seek updates.
      84d0eb4f
    • Thomas Vander Stichele's avatar
      style fixes · 3cbcad1d
      Thomas Vander Stichele authored
      Original commit message from CVS:
      style fixes
      3cbcad1d
    • Andy Wingo Wingo's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c · 7afb1045
      Andy Wingo Wingo authored
      Original commit message from CVS:
      2005-08-23  Andy Wingo  <wingo@pobox.com>
      
      * gst-libs/gst/audio/gstbaseaudiosrc.c
      (gst_base_audio_src_fixate): Only fixate endianness if it is
      present in the caps.
      7afb1045