1. 12 Apr, 2007 5 commits
    • Wim Taymans's avatar
      gst/videorate/gstvideorate.c: Add some debug. · 807258cc
      Wim Taymans authored
      Original commit message from CVS:
      * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
      (gst_video_rate_chain):
      Add some debug.
      * tests/check/elements/videorate.c: (GST_START_TEST),
      (videorate_suite):
      Added check for videorate changing caps handling. Closes #421834.
      807258cc
    • Michael Smith's avatar
      ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating... · cda0d2dc
      Michael Smith authored
      ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
      
      Original commit message from CVS:
      * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
      Use scale functions to avoid overflow when calculating duration of
      vorbis buffers.
      cda0d2dc
    • Tim-Philipp Müller's avatar
      API: add gst_tag_freeform_string_to_utf8() (#405072). · a2084690
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * gst-libs/gst/tag/tag.h:
      * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
      API: add gst_tag_freeform_string_to_utf8() (#405072).
      * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
      Use gst_tag_freeform_string_to_utf8() here.
      a2084690
    • Thomas Vander Stichele's avatar
      log tweaking · 8a6b8cfb
      Thomas Vander Stichele authored
      Original commit message from CVS:
      log tweaking
      8a6b8cfb
    • Wim Taymans's avatar
      gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly. · 3e455f8a
      Wim Taymans authored
      Original commit message from CVS:
      * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
      (gst_gdp_pay_sink_event):
      Make sure we set the IN_CAPS flag correctly.
      * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
      Get the IN_CAPS flag before we call functions that mess with the flags.
      3e455f8a
  2. 10 Apr, 2007 3 commits
  3. 06 Apr, 2007 2 commits
  4. 05 Apr, 2007 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration... · b802dea8
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_skew_slaving):
      Don't try to create invalid calibration parameters by making the
      internal time go backwards, instead make external time go forward.
      b802dea8
    • Tommi Myöhänen's avatar
      gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when... · 32a72762
      Tommi Myöhänen authored
      gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
      
      Original commit message from CVS:
      Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
      * gst/playback/gstplaybasebin.c: (add_stream):
      Fix leak in add_stream(), when g_value_set_object() increases the
      refcount of streaminfo object. Fixes #426250.
      32a72762
  5. 04 Apr, 2007 1 commit
    • David Schleef's avatar
      gst/videotestsrc/: Add a test pattern called "circular", which has concentric... · e859791a
      David Schleef authored
      gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency.  T...
      
      Original commit message from CVS:
      * gst/videotestsrc/gstvideotestsrc.c:
      * gst/videotestsrc/gstvideotestsrc.h:
      * gst/videotestsrc/videotestsrc.c:
      * gst/videotestsrc/videotestsrc.h:
      Add a test pattern called "circular", which has concentric
      rings with varying radial frequency.  The main purpose of this
      pattern is to test fidelity loss in a filter or scaler element.
      Notably, this pattern is scale invariant, and is optimally viewed
      with a width (and height) of 400.
      e859791a
  6. 03 Apr, 2007 1 commit
    • Tommi Myöhänen's avatar
      gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions: · 8676f3dc
      Tommi Myöhänen authored
      Original commit message from CVS:
      Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
      * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
      (deactivate_free_recursive):
      Decodebin2 doesn't unref pads it obtains in some occasions:
      - multiqueue src pads, when either connecting further or exposing
      - sink pads of new autoplugged elements
      - peer pads when recursively freeing elements
      Fixes #425455.
      8676f3dc
  7. 30 Mar, 2007 2 commits
  8. 29 Mar, 2007 7 commits
    • René Stadler's avatar
      with some minor changes · 6ac8ff9e
      René Stadler authored
      Original commit message from CVS:
      Patch by: René Stadler <mail at renestadler dot de>
      with some minor changes
      * gst-libs/gst/floatcast/floatcast.h:
      Use more efficient float endianness conversion functions that don't
      involve 2 function calls per value.
      * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
      (check_default), (audio_convert_prepare_context):
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_parse_caps), (make_lossless_changes):
      Support non-native endianness floats as input and output.
      Fixes #339838.
      * tests/check/elements/audioconvert.c: (verify_convert),
      (GST_START_TEST):
      Add unit tests for the non-native endianness float conversions.
      6ac8ff9e
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure. · 76462ceb
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_base_init),
      (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
      (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
      (gst_base_rtp_depayload_set_gst_timestamp),
      (gst_base_rtp_depayload_change_state),
      (gst_base_rtp_depayload_set_property),
      (gst_base_rtp_depayload_get_property):
      * gst-libs/gst/rtp/gstbasertpdepayload.h:
      Add Private structure.
      Bring element code to 2007.
      Parse clock-base caps param and use it when generating the
      newsegment.
      Reset variables before going to PAUSED.
      Fix some docs.
      76462ceb
    • Wim Taymans's avatar
      Add RTCP docs. · 0a39f494
      Wim Taymans authored
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-docs.sgml:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
      (gst_base_rtp_audio_payload_get_adapter):
      Add RTCP docs.
      Fix some more docs.
      * gst-libs/gst/rtp/Makefile.am:
      * gst-libs/gst/rtp/gstrtcpbuffer.c:
      (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
      (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
      (gst_rtcp_buffer_get_packet_count), (read_packet_header),
      (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
      (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
      (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
      (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
      (gst_rtcp_packet_sr_get_sender_info),
      (gst_rtcp_packet_sr_set_sender_info),
      (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
      (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
      (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
      (gst_rtcp_packet_sdes_get_chunk_count),
      (gst_rtcp_packet_sdes_first_chunk),
      (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
      (gst_rtcp_packet_sdes_first_item),
      (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
      (gst_rtcp_packet_bye_get_ssrc_count),
      (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
      (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
      (gst_rtcp_packet_bye_get_reason_len),
      (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
      * gst-libs/gst/rtp/gstrtcpbuffer.h:
      Add new helper object for parsing and creating RTCP messages.
      0a39f494
    • Sebastian Dröge's avatar
      gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always... · dfdd873f
      Sebastian Dröge authored
      gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
      
      Original commit message from CVS:
      * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
      PCM samples with width=8 must be always unsigned, no matter what
      depth they have.
      dfdd873f
    • Andy Wingo's avatar
      gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets... · af17f81a
      Andy Wingo authored
      gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
      
      Original commit message from CVS:
      2007-03-29  Andy Wingo  <wingo@pobox.com>
      
      * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
      perfect offsets also, not just timestamps.
      
      * tests/check/elements/videorate.c (test_more): Test that given
      any incoming offsets, that videorate produces perfect offsets.
      af17f81a
    • Wim Taymans's avatar
      gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats. · d4015266
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/riff/riff-ids.h:
      Add some more RIFF formats.
      d4015266
    • 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
  9. 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
  10. 27 Mar, 2007 4 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
    • Thomas Vander Stichele's avatar
      commit new file · a6457e16
      Thomas Vander Stichele authored
      Original commit message from CVS:
      commit new file
      a6457e16
  11. 26 Mar, 2007 3 commits
  12. 23 Mar, 2007 2 commits
    • Christian Schaller's avatar
      update spec file · f58ea201
      Christian Schaller authored
      Original commit message from CVS:
      update spec file
      f58ea201
    • 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
  13. 22 Mar, 2007 3 commits
  14. 21 Mar, 2007 2 commits
  15. 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