1. 22 May, 2008 3 commits
    • Jan Schmidt's avatar
      Add some documentation comments, and some new headers to be scanned. · d58def62
      Jan Schmidt authored
      Original commit message from CVS:
      * docs/plugins/Makefile.am:
      * docs/plugins/gst-plugins-base-plugins-overrides.txt:
      * docs/plugins/gst-plugins-base-plugins-sections.txt:
      * docs/plugins/gst-plugins-base-plugins.args:
      * docs/plugins/gst-plugins-base-plugins.hierarchy:
      * docs/plugins/gst-plugins-base-plugins.interfaces:
      * docs/plugins/gst-plugins-base-plugins.prerequisites:
      * docs/plugins/inspect/plugin-adder.xml:
      * docs/plugins/inspect/plugin-alsa.xml:
      * docs/plugins/inspect/plugin-audioconvert.xml:
      * docs/plugins/inspect/plugin-audiorate.xml:
      * docs/plugins/inspect/plugin-audioresample.xml:
      * docs/plugins/inspect/plugin-audiotestsrc.xml:
      * docs/plugins/inspect/plugin-cdparanoia.xml:
      * docs/plugins/inspect/plugin-decodebin.xml:
      * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
      * docs/plugins/inspect/plugin-gdp.xml:
      * docs/plugins/inspect/plugin-gio.xml:
      * docs/plugins/inspect/plugin-gnomevfs.xml:
      * docs/plugins/inspect/plugin-libvisual.xml:
      * docs/plugins/inspect/plugin-ogg.xml:
      * docs/plugins/inspect/plugin-pango.xml:
      * docs/plugins/inspect/plugin-playback.xml:
      * docs/plugins/inspect/plugin-queue2.xml:
      * docs/plugins/inspect/plugin-subparse.xml:
      * docs/plugins/inspect/plugin-tcp.xml:
      * docs/plugins/inspect/plugin-theora.xml:
      * docs/plugins/inspect/plugin-typefindfunctions.xml:
      * docs/plugins/inspect/plugin-uridecodebin.xml:
      * docs/plugins/inspect/plugin-video4linux.xml:
      * docs/plugins/inspect/plugin-videorate.xml:
      * docs/plugins/inspect/plugin-videoscale.xml:
      * docs/plugins/inspect/plugin-videotestsrc.xml:
      * docs/plugins/inspect/plugin-volume.xml:
      * docs/plugins/inspect/plugin-vorbis.xml:
      * docs/plugins/inspect/plugin-ximagesink.xml:
      * docs/plugins/inspect/plugin-xvimagesink.xml:
      * ext/cdparanoia/gstcdparanoiasrc.c:
      * ext/ogg/gstoggdemux.c:
      * ext/ogg/gstoggdemux.h:
      * ext/ogg/gstoggmux.c:
      * ext/ogg/gstoggmux.h:
      * gst/audioconvert/audioconvert.c:
      * gst/audioconvert/audioconvert.h:
      * gst/audioconvert/gstaudioconvert.h:
      * gst/gdp/gstgdpdepay.h:
      * gst/gdp/gstgdppay.h:
      * gst/playback/gstdecodebin.c:
      * gst/playback/gstdecodebin2.c:
      * gst/playback/gstplaybin.c:
      * gst/playback/gstplaybin2.c:
      * gst/playback/gsturidecodebin.c:
      * gst/tcp/gstmultifdsink.c:
      * gst/tcp/gstmultifdsink.h:
      * gst/tcp/gsttcp.h:
      Add some documentation comments, and some new headers to be scanned.
      Rename some internal enum declarations (audioconvert's DitherType and
      NoiseShapingType, GstUnitType from the TCP elements) to match the
      documented GObject type names so that the docs pick them up.
      Name the playbin2 docs markups properly so they get picked up. They'll
      need renaming back when/if playbin2 becomes playbin.
      100% symbol coverage for the plugin docs, booya.
      d58def62
    • Thijs Vermeir's avatar
      gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454. · 88b1e8ef
      Thijs Vermeir authored
      Original commit message from CVS:
      Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
      * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
      Fix generation of NV12/NV21 frames. Fixes bug #532454.
      88b1e8ef
    • Sjoerd Simons's avatar
      gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to... · 1c424d9d
      Sjoerd Simons authored
      gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
      
      Original commit message from CVS:
      Patch by: Sjoerd Simons <sjoerd at luon dot net>
      * gst/playback/gstdecodebin.c: (remove_fakesink):
      Lock the fakesink before setting the state to NULL and removing it from
      the bin so that a concurrent state change cannot interfere.
      Fixes #534331.
      1c424d9d
  2. 21 May, 2008 16 commits
    • Felipe Contreras's avatar
      docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled. · 75d05dc4
      Felipe Contreras authored
      Original commit message from CVS:
      * docs/Makefile.am:
      Fix installing plugin documentation when gtk-doc is disabled.
      75d05dc4
    • Felipe Contreras's avatar
      gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h · b5f896da
      Felipe Contreras authored
      Original commit message from CVS:
      * gst-libs/gst/rtsp/Makefile.am:
      Distribute, don't install md5.h
      b5f896da
    • Julien Moutte Moutte's avatar
      gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms. · 0f80e462
      Julien Moutte Moutte authored
      Original commit message from CVS:
      2008-05-21  Julien Moutte  <julien@fluendo.com>
      
      * gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
      instead of SOL_IP, works on more platforms.
      * gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
      arguments.
      0f80e462
    • Wim Taymans's avatar
      Some debug and comment fixes. · 2cdf18ed
      Wim Taymans authored
      Original commit message from CVS:
      * ext/vorbis/vorbisdec.c:
      * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
      * sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
      Some debug and comment fixes.
      * tests/examples/dynamic/addstream.c: (main):
      Fix , to ;
      2cdf18ed
    • Wim Taymans's avatar
      Don't use bad gst_element_get_pad(). · c6b54c3d
      Wim Taymans authored
      Original commit message from CVS:
      * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
      * gst/playback/decodetest.c: (new_decoded_pad_cb):
      * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
      (try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
      (cleanup_decodebin):
      * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
      (connect_element), (gst_decode_group_control_demuxer_pad):
      * gst/playback/gstplaybasebin.c: (queue_remove_probe),
      (queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
      (mute_group_type):
      * gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
      (gst_play_bin_set_property), (handoff), (gen_video_element),
      (gen_text_element), (gen_audio_element), (gen_vis_element),
      (remove_sinks), (add_sink), (setup_sinks):
      * gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
      * gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
      (gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
      (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
      (gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
      (gen_video_chain), (gen_text_chain), (gen_audio_chain),
      (gen_vis_chain), (gst_play_sink_reconfigure),
      (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
      (gst_play_sink_request_pad):
      * gst/playback/gsturidecodebin.c: (type_found), (setup_source):
      * gst/playback/test.c: (gen_video_element), (gen_audio_element),
      (cb_newpad):
      * gst/playback/test6.c: (new_decoded_pad_cb):
      * tests/check/elements/audioconvert.c: (GST_START_TEST):
      * tests/check/elements/audiorate.c: (test_injector_chain),
      (do_perfect_stream_test):
      * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
      * tests/check/elements/gdpdepay.c: (GST_START_TEST):
      * tests/check/elements/gnomevfssink.c:
      * tests/check/elements/textoverlay.c:
      (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
      * tests/check/elements/videotestsrc.c: (GST_START_TEST):
      * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
      * tests/check/pipelines/oggmux.c: (test_pipeline):
      * tests/check/pipelines/streamheader.c: (GST_START_TEST):
      * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
      * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
      * tests/examples/seek/scrubby.c: (make_wav_pipeline):
      * tests/examples/seek/seek.c: (make_mod_pipeline),
      (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
      (make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
      (make_theora_pipeline), (make_vorbis_theora_pipeline),
      (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
      (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
      (update_fill), (msg_buffering):
      Don't use bad gst_element_get_pad().
      c6b54c3d
    • Stefan Kost's avatar
      gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation... · eda6d89b
      Stefan Kost authored
      gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
      
      Original commit message from CVS:
      * gst-libs/gst/riff/riff-media.c:
      Fix wrong method name in docs. Fix calculation of strf fields for
      broken mulaw/alaw.
      * gst-libs/gst/riff/riff-read.c:
      Whitespace fix and removing double ';'.
      eda6d89b
    • Wim Taymans's avatar
      docs/design/part-playbin2.txt: Add some leftover doc. · 3cd156ca
      Wim Taymans authored
      Original commit message from CVS:
      * docs/design/part-playbin2.txt:
      Add some leftover doc.
      3cd156ca
    • Sebastian Dröge's avatar
      gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit. · 736b1819
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
      Fix copy & paste error in last commit.
      736b1819
    • Sebastian Dröge's avatar
      gst/audioconvert/gstchannelmix.c: Add support for mixing... · 7d605d45
      Sebastian Dröge authored
      gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
      
      Original commit message from CVS:
      * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
      Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
      other channel positions when source has SIDE channels and dest doesn't
      or the other way around.
      7d605d45
    • Henrik Eriksson's avatar
      gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933. · 10ae17ce
      Henrik Eriksson authored
      Original commit message from CVS:
      Patch by: Henrik Eriksson <henriken at axis dot com>
      * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
      (gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
      (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
      (gst_multi_fd_sink_get_property):
      * gst/tcp/gstmultifdsink.h:
      Add support for DSCP QOS. Fixes #469933.
      10ae17ce
    • Sebastian Dröge's avatar
      tests/check/elements/audioconvert.c: Add another test that checks if... · 74d46a99
      Sebastian Dröge authored
      tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
      
      Original commit message from CVS:
      * tests/check/elements/audioconvert.c: (GST_START_TEST):
      Add another test that checks if conversion between standard 1 and 2
      channel layouts with and without positions set is working.
      74d46a99
    • Sebastian Dröge's avatar
      gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts. · d03bbd1e
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst-libs/gst/audio/multichannel.c:
      (gst_audio_check_channel_positions):
      Allow non-standard 2 channel layouts.
      * tests/check/elements/audioconvert.c: (GST_START_TEST):
      Add some tests for converting and remapping non-standard 1 and 2
      channel layouts.
      d03bbd1e
    • Sebastian Dröge's avatar
      gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix... · d47bd6d7
      Sebastian Dröge authored
      gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
      
      Original commit message from CVS:
      * gst/audioconvert/gstchannelmix.c:
      (gst_channel_mix_fill_normalize):
      Prevent division by zero if the channel mix matrix contains only
      zeroes.
      d47bd6d7
    • Antoine Tremblay's avatar
      gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071. · a8dda35c
      Antoine Tremblay authored
      Original commit message from CVS:
      Patch by: Antoine Tremblay <hexa00 at gmail dot com>
      * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
      Close a buffer memory leak. Fixes bug #534071.
      a8dda35c
    • Sebastian Dröge's avatar
      gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members... · 3ee2676c
      Sebastian Dröge authored
      gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
      
      Original commit message from CVS:
      * gst-libs/gst/rtsp/gstrtsptransport.h:
      Make the GstRTSPTransport struct members public as there are no
      setters/getters and it's supposed to be changed directly.
      Fixes bug #533087.
      3ee2676c
    • Sebastian Dröge's avatar
      gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with... · e66b0a66
      Sebastian Dröge authored
      gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth so don't claim this on the pad tem...
      
      Original commit message from CVS:
      * gst/adder/gstadder.c:
      Adder also doesn't support audio/x-raw-int with width!=depth so don't
      claim this on the pad template caps.
      e66b0a66
  3. 20 May, 2008 7 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration... · f36d9d6b
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_sync_latency):
      We can only use our optimal calibration if we prerolled before the
      latency expired.
      f36d9d6b
    • Tim-Philipp Müller's avatar
      configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic. · d0932b0a
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * configure.ac:
      Require core CVS for GstBaseSrc buffer caps setting magic.
      d0932b0a
    • Sebastian Dröge's avatar
      gst/audioconvert/gstaudioconvert.c: Fix logic in last commit. · fcda3964
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_fixate_channels):
      Fix logic in last commit.
      fcda3964
    • Sebastian Dröge's avatar
      gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the... · d76c4b4c
      Sebastian Dröge authored
      gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
      
      Original commit message from CVS:
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_fixate_channels):
      Passthrough the channel positions if the number of output channels is
      the same as the number of input channels, the input had a channel
      layout and downstream requests no special one. We did this already for
      > 2 channels but now it's also done for 1 channel. Fixes bug #533617.
      d76c4b4c
    • Wim Taymans's avatar
      ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they... · d8dc371c
      Wim Taymans authored
      ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
      
      Original commit message from CVS:
      * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
      (gst_gnome_vfs_src_finalize),
      (gst_gnome_vfs_src_received_headers_callback),
      (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
      * ext/gnomevfs/gstgnomevfssrc.h:
      Set the ICY caps on the srcpad from where they get picked up by the base
      class now and set on the outgoing buffers.
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create):
      * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
      BaseSrc now sets the caps on outgoing buffers automatically.
      d8dc371c
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer... · 95d162fb
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_resample_slaving),
      (gst_base_audio_sink_skew_slaving),
      (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
      (gst_base_audio_sink_async_play),
      (gst_base_audio_sink_change_state):
      Change the way in which the ringbuffer is started when dealing with a
      slaved clock and latency. We now sync to the clock until we reach
      upstream latency before starting the ringbuffer. This has the effect
      that we can accurately align the master and slave clocks and let the
      rate correction code take care of the initial drift or rounding errors
      instead of leaving them uncorrected with the old approach.
      95d162fb
    • Sebastian Dröge's avatar
      gst/audioconvert/gstaudioconvert.c: Correctly set the default channel... · b5a5d647
      Sebastian Dröge authored
      gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
      
      Original commit message from CVS:
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_fixate_channels):
      Correctly set the default channel positions when converting to 8
      channels.
      b5a5d647
  4. 19 May, 2008 3 commits
  5. 16 May, 2008 1 commit
  6. 14 May, 2008 5 commits
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further... · 86ab5120
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_chain):
      Validate the RTP packet before further processing it. It's just too
      dangerous to accept random packets and people are not forced to use a
      jitterbuffer or session manager to filter out the bad packets.
      * gst-libs/gst/rtp/gstrtpbuffer.c:
      (gst_rtp_buffer_set_extension_data),
      (gst_rtp_buffer_get_payload_subbuffer):
      Small cleanups.
      When setting extension data in a buffer that is too small, we fail and
      we should not set the extension bit.
      Change GST_WARNINGS into g_warning because they really are
      programming errors.
      * tests/check/libs/rtp.c: (GST_START_TEST):
      Catch the g_warnings now in the unit tests and that fact that failing to
      set extension data left the extension bit untouched.
      86ab5120
    • Tim-Philipp Müller's avatar
      gst/audioresample/gstaudioresample.c: Revert previous change which made... · d92ff26d
      Tim-Philipp Müller authored
      gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
      
      Original commit message from CVS:
      * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
      Revert previous change which made basetransform handle buffer_alloc
      and which breaks things badly in the non-passthrough case since it
      returned buffers with a different (ie. sometimes smaller) size than
      the size requested.
      d92ff26d
    • Bernard B's avatar
      gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase... · d06df554
      Bernard B authored
      gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
      
      Original commit message from CVS:
      Patch by: Bernard B <b-gnome at largestprime dot net>
      * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
      Fix seqnum compare function for bordercase values and fix the docs
      again. Fixes #533075.
      * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
      Add a testcase for seqnum compare function.
      d06df554
    • Sebastian Dröge's avatar
      gst/adder/gstadder.c: Correctly declare the supported endianness on the pad... · 6720c5be
      Sebastian Dröge authored
      gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
      
      Original commit message from CVS:
      * gst/adder/gstadder.c: (gst_adder_setcaps),
      (gst_adder_class_init):
      Correctly declare the supported endianness on the pad templates
      and check for correct endianness in the set caps function. Adder
      only supports native endianness.
      Also use gst_element_class_set_details_simple().
      6720c5be
    • Stefan Kost's avatar
      sys/xvimage/xvimagesink.c: Better debug logging in port value handling.... · 5965f5e8
      Stefan Kost authored
      sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.
      
      Original commit message from CVS:
      * sys/xvimage/xvimagesink.c:
      Better debug logging in port value handling. Merging separate port
      value loops into one.
      5965f5e8
  7. 13 May, 2008 5 commits