1. 03 Sep, 2005 1 commit
  2. 02 Sep, 2005 7 commits
  3. 31 Aug, 2005 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Resync if the buffer timestamps drift... · 44cc3421
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Resync if the buffer timestamps drift more than a 10th of a second.
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render):
      Resync if the buffer timestamps drift more than a 10th
      of a second.
      44cc3421
    • Tim-Philipp Müller's avatar
      sys/v4l/gstv4lsrc.c: The 'timestamp-offset' property is registered as an... · 13a09b13
      Tim-Philipp Müller authored
      sys/v4l/gstv4lsrc.c: The 'timestamp-offset' property is registered as an int64, so let's use g_value_{set|get}_int64(...
      
      Original commit message from CVS:
      * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_set_property),
      (gst_v4lsrc_get_property):
      The 'timestamp-offset' property is registered as an int64, so
      let's use g_value_{set|get}_int64() in our setter and getter
      functions (makes it work and fixes warnings with gst-inspect).
      13a09b13
  4. 30 Aug, 2005 3 commits
    • Wim Taymans's avatar
      check/elements/: Fix checks. · 0b18cb8f
      Wim Taymans authored
      Original commit message from CVS:
      * check/elements/audioconvert.c: (setup_audioconvert):
      * check/elements/audioresample.c: (setup_audioresample):
      * check/elements/volume.c: (setup_volume):
      Fix checks.
      0b18cb8f
    • Thomas Vander Stichele's avatar
      make module a param · 5ea209dd
      Thomas Vander Stichele authored
      Original commit message from CVS:
      * common/gtk-doc-plugins.mak:
      * common/plugins.xsl:
      * docs/plugins/Makefile.am:
      make module a param
      5ea209dd
    • Stefan Kost's avatar
      examples/seeking/seek.c: update the example · 85056f97
      Stefan Kost authored
      Original commit message from CVS:
      * examples/seeking/seek.c: (make_mp3_pipeline),
      (make_mpeg_pipeline), (seek_cb), (start_seek), (stop_seek),
      (play_cb), (pause_cb), (stop_cb):
      update the example
      85056f97
  5. 29 Aug, 2005 9 commits
  6. 28 Aug, 2005 1 commit
  7. 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
  8. 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