1. 29 Mar, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs. · 804e7d17
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstrtpbuffer.c:
      (gst_rtp_buffer_default_clock_rate):
      * gst-libs/gst/rtp/gstrtpbuffer.h:
      Fix fixed payload names and docs.
      Added method to get the default clock rates of fixed payload types.
      API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
      804e7d17
  2. 28 Mar, 2007 2 commits
    • Zaheer Abbas Merali's avatar
      tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore. · 01038e30
      Zaheer Abbas Merali authored
      Original commit message from CVS:
      * tests/check/pipelines/.cvsignore:
      Add new vorbisdec test to cvsignore.
      01038e30
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate. · 450030eb
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
      (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
      (gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
      (gst_base_audio_sink_set_property),
      (gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
      (clock_convert_external), (gst_base_audio_sink_resample_slaving),
      (gst_base_audio_sink_skew_slaving),
      (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
      (gst_base_audio_sink_async_play):
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      Store private stuff in GstBaseAudioSinkPrivate.
      Add configurable clock slaving modes property.
      API:: GstBaseAudioSink::slave-method property
      Some more latency reporting tweaks.
      Added skew based clock slaving correction and make it the default until
      the resampling method is more robust.
      450030eb
  3. 27 Mar, 2007 3 commits
    • Sebastian Dröge's avatar
      gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and... · 293a9c09
      Sebastian Dröge authored
      gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
      
      Original commit message from CVS:
      * gst/audioconvert/audioconvert.c:
      Add docs to the integer pack functions and implement proper
      rounding. Before we had rounding towards negative infinity, i.e.
      always the smaller number was taken. Now we use natural rounding,
      i.e. rounding to the nearest integer and to the one with the largest
      absolute value for X.5. The old rounding introduced some minor
      distortions. Fixes #420079
      * tests/check/elements/audioconvert.c: (GST_START_TEST):
      Fix one unit test that assumed the old rounding and added unit tests
      for checking signed/unsigned int16 <-> signed/unsigned int16 with
      depth 8, one for signed int16 <-> unsigned int16 and one for the new
      rounding from signed int32 to signed/unsigned int16.
      293a9c09
    • Michael Smith's avatar
      gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced... · e1544977
      Michael Smith authored
      gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
      
      Original commit message from CVS:
      * gst/audioconvert/gstaudioconvert.c: (strip_width_64),
      (gst_audio_convert_transform_caps):
      Fix typo in debug line introduced recently, as pointed out on irc.
      e1544977
    • Tim-Philipp Müller's avatar
      Make sure we parse floating-point numbers in vorbis comments correctly with... · 726f2c17
      Tim-Philipp Müller authored
      Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
      
      Original commit message from CVS:
      * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
      * tests/check/libs/tag.c: (GST_START_TEST):
      Make sure we parse floating-point numbers in vorbis comments
      correctly with either '.' or ',' as separator, no matter what
      the current locale is. Add unit test for this too.
      726f2c17
  4. 26 Mar, 2007 3 commits
  5. 23 Mar, 2007 1 commit
    • Michael Smith's avatar
      gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use... · b3827533
      Michael Smith authored
      gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
      
      Original commit message from CVS:
      * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
      (gst_video_rate_reset), (gst_video_rate_chain):
      If videorate changes caps, we can no longer use the old buffer
      (which may have a different size, incompatible with our caps).
      So don't do that; just duplicate the new frame more times.
      b3827533
  6. 22 Mar, 2007 2 commits
  7. 21 Mar, 2007 2 commits
  8. 20 Mar, 2007 1 commit
    • Michael Smith's avatar
      ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to... · 45b6d734
      Michael Smith authored
      ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
      
      Original commit message from CVS:
      * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
      If we get a zero-sized input buffer, don't pass it to libvorbis, as
      that marks EOS internally. After that, libvorbis will buffer all
      input data, and encode none of it, eventually leading to memory
      exhaustion.
      45b6d734
  9. 19 Mar, 2007 1 commit
    • Wim Taymans's avatar
      gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore. · d24780a0
      Wim Taymans authored
      Original commit message from CVS:
      * gst/playback/gstdecodebin.c: (remove_fakesink):
      Don't post STATE_DIRTY anymore.
      * gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
      (gst_play_bin_change_state):
      Remove stream_time reset in seek handling, core does that now.
      Disable clocking for live pipelines by forcing a NULL clock to the
      complete pipeline, core is too smart now for our previous hack.
      We can always autoplug in PAUSED now.
      d24780a0
  10. 18 Mar, 2007 1 commit
  11. 16 Mar, 2007 2 commits
    • Michael Smith's avatar
      gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke... · 3bc107dd
      Michael Smith authored
      gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
      
      Original commit message from CVS:
      * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
      (strip_width_64), (append_with_other_format):
      Previous fix was too simplistic, and broke the tests. Use a better
      approach; only strip 64 from widths for integer audio.
      3bc107dd
    • Michael Smith's avatar
      gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so... · 5759241e
      Michael Smith authored
      gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
      
      Original commit message from CVS:
      * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
      (gst_audio_convert_transform_caps):
      We don't support 64 bit integer audio, so don't try to claim we can.
      Stops us producing caps don't match our template caps.
      Update comments.
      5759241e
  12. 15 Mar, 2007 1 commit
    • Michael Smith's avatar
      gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very... · 4ab2d699
      Michael Smith authored
      gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
      
      Original commit message from CVS:
      * gst/audioresample/gstaudioresample.c:
      (audioresample_check_discont), (audioresample_transform):
      Don't trigger discontinuities for very small imperfections; a filter
      flush will sound bad, and many plugins have rounding errors leading
      to these.
      4ab2d699
  13. 14 Mar, 2007 4 commits
    • Philippe Kalaf's avatar
      gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk. · b6d7f654
      Philippe Kalaf authored
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
      * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
      Add min-ptime property to RTP base audio payloader. Patch by
      olivier.crete@collabora.co.uk.
      Fixes #415001
      
      Indentation/whitespace/documentation fixes.
      b6d7f654
    • Julien Moutte's avatar
      gst/audioresample/gstaudioresample.c: Handle discontinuous streams. · 6940042e
      Julien Moutte authored
      Original commit message from CVS:
      2007-03-14  Julien MOUTTE  <julien@moutte.net>
      
      * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
      (audioresample_transform_size), (audioresample_do_output),
      (audioresample_transform), (audioresample_pushthrough): Handle
      discontinuous streams.
      * gst/audioresample/gstaudioresample.h:
      * tests/check/elements/audioresample.c:
      (test_discont_stream_instance), (GST_START_TEST),
      (audioresample_suite): Add a test for discontinuous streams.
      * win32/common/config.h: Updated.
      6940042e
    • Thomas Vander Stichele's avatar
      po/: Update translations from translation project. · 8b80c6f1
      Thomas Vander Stichele authored
      Original commit message from CVS:
      * po/af.po:
      * po/az.po:
      * po/cs.po:
      * po/en_GB.po:
      * po/it.po:
      * po/nb.po:
      * po/nl.po:
      * po/or.po:
      * po/sq.po:
      * po/sr.po:
      * po/sv.po:
      * po/uk.po:
      * po/vi.po:
      Update translations from translation project.
      8b80c6f1
    • Thomas Vander Stichele's avatar
      gst/audioresample/: Since I really am not interested in a debug line for each... · 081deac0
      Thomas Vander Stichele authored
      gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
      
      Original commit message from CVS:
      * gst/audioresample/debug.h:
      * gst/audioresample/resample.c: (resample_init):
      Since I really am not interested in a debug line for each sample
      being processed, move the library's debugging to its own category,
      libaudioresample
      081deac0
  14. 12 Mar, 2007 2 commits
    • Michael Smith's avatar
      ext/theora/theoradec.c: Since the plugin doesn't support anything other than... · 72929734
      Michael Smith authored
      ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
      
      Original commit message from CVS:
      * ext/theora/theoradec.c: (theora_handle_type_packet):
      Since the plugin doesn't support anything other than 4:2:0 right
      now, post an error and fail if we get something else. Won't matter
      until libtheora supports the other pixel formats, but hopefully
      that'll be soon...
      72929734
    • Alex Lancaster's avatar
      I'm too lazy to comment this · 7931b94d
      Alex Lancaster authored
      Original commit message from CVS:
      Mention Patch by: Alex Lancaster in a recent commit.
      7931b94d
  15. 10 Mar, 2007 4 commits
    • Sébastien Moutte's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion. · 1596dd26
      Sébastien Moutte authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
      Use gst_guint64_to_gdouble for conversion.
      * win32/MANIFEST:
      Add new files to the win32 MANIFEST.
      * win32/common/libgstaudio.def:
      * win32/common/libgstpbutils.def:
      Add new exported functions.
      * win32/vs6/gst_plugins_base.dsw:
      * win32/vs6/libgstdecodebin.dsp:
      * win32/vs6/libgstplaybin.dsp:
      Change the link to libgstpbutils.lib.
      * win32/vs6/libgstdecodebin2.dsp:
      Add a new project for decodebin2.
      * win32/vs6/libgstpbutils.dsp:
      Add a new project for pbutils.
      1596dd26
    • Tim-Philipp Müller's avatar
      gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and... · 4462906b
      Tim-Philipp Müller authored
      gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
      
      Original commit message from CVS:
      * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
      Also accept partial dates with only year and month,
      like 1999-12-00 (fixes #410396 even more).
      * tests/check/libs/tag.c: (GST_START_TEST):
      Add unit test for the above.
      4462906b
    • Tim-Philipp Müller's avatar
      tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799). · 5de913c2
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * tests/check/elements/subparse.c: (GST_START_TEST),
      (subparse_suite):
      Add unit test for MPL2 subtitle format (#413799).
      5de913c2
    • Kamil Pawlowski's avatar
      gst/subparse/: Add support for MPL2 subtitle format (#413799). · 389eb6f2
      Kamil Pawlowski authored
      Original commit message from CVS:
      Patch by: Kamil Pawlowski  <kamilpe gmail com>
      * gst/subparse/Makefile.am:
      * gst/subparse/gstsubparse.c:
      (gst_sub_parse_data_format_autodetect),
      (gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
      (gst_subparse_type_find):
      * gst/subparse/gstsubparse.h:
      * gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
      * gst/subparse/mpl2parse.h:
      Add support for MPL2 subtitle format (#413799).
      389eb6f2
  16. 09 Mar, 2007 9 commits
  17. 08 Mar, 2007 1 commit