1. 31 May, 2008 1 commit
    • Mark Nauwelaerts's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and... · c660bbd6
      Mark Nauwelaerts authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
      (gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
      (gst_base_audio_src_change_state):
      Provide readable actual-buffer-time and actual-latency-time properties
      that reflect the configured ringbuffer values. Fixes #524724.
      c660bbd6
  2. 30 May, 2008 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into... · 11309247
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
      (gst_basertppayload_change_state):
      Simply converting the running time into an RTP timestamp by scaling it
      based on the clock-rate is good enough for making an RTP timestamp. This
      has the added benefit that we can later on expose a property with the
      RTP timestamp of running time 0, as is needed for RTSP servers to
      generate the response of the PLAY request.
      11309247
    • Sebastian Dröge's avatar
      gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now... · fdd708c4
      Sebastian Dröge authored
      gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
      
      Original commit message from CVS:
      * gst/audioconvert/gstaudioconvert.c:
      (structure_has_fixed_channel_positions),
      (gst_audio_convert_transform_caps):
      Allow up to 11 positioned channels now that audioconvert can handle
      this but add no default positions for > 8 channels.
      * tests/check/elements/audioconvert.c: (GST_START_TEST):
      Add some unit tests for the above change: Test conversion of
      11 positioned channels to stereo and the other way around, test
      conversion of 15 unpositioned channels in different ways.
      fdd708c4
  3. 29 May, 2008 5 commits
    • Sebastian Dröge's avatar
      win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols. · 79cf1cf8
      Sebastian Dröge authored
      Original commit message from CVS:
      * win32/common/libgstaudio.def:
      Add gst_audio_clock_reset to the list of exported symbols.
      79cf1cf8
    • Sebastian Dröge's avatar
      tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header... · ca7a0b8e
      Sebastian Dröge authored
      tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
      
      Original commit message from CVS:
      * tests/check/elements/vorbisdec.c: (vorbisdec_suite):
      Remove wrong_channels_identification_header unit test as we now
      support 7 (and more channels).
      ca7a0b8e
    • Sebastian Dröge's avatar
      gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the... · b86a5d43
      Sebastian Dröge authored
      gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
      
      Original commit message from CVS:
      * gst/audioconvert/gstchannelmix.c:
      (gst_channel_mix_fill_one_other):
      If mixing left or right to center (or the other way around) only take
      the complete value if we don't already have the original position in
      the source.
      b86a5d43
    • Sebastian Dröge's avatar
      gst-libs/gst/audio/multichannel.c: Allow rear center together with rear... · 45ef6b5e
      Sebastian Dröge authored
      gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/multichannel.c:
      (gst_audio_check_channel_positions),
      (gst_audio_set_structure_channel_positions_list),
      (gst_audio_fixate_channel_positions):
      Allow rear center together with rear left/right and other previously
      conflicting channel positions. The reason why they weren't allowed
      was the channel mixing implementation in audioconvert.
      Also take this into account when fixing channel layouts.
      Allow setting channel positions for 1/2 channels when using
      gst_audio_set_structure_channel_position().
      * gst/audioconvert/gstchannelmix.c:
      (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_special), (gst_channel_mix_fill_matrix):
      Major rewrite of the channel mixing.
      We now allow previously	conflicting channel positions to appear
      together (rear center and rear left/right for example).
      Fixes bug #533817.
      Rework the way channels are mixed together to take more possible
      channel positions into account, properly mix from/to side channels
      and don't assume that either center, left&right or nothing of a
      specific position is available anymore.
      * tests/check/elements/audioconvert.c: (GST_START_TEST):
      Adjust unit tests with non-standard 1/2 channel layouts to the more
      correct new behaviour.
      Add a unit test for 5.1->Stereo downmixing.
      45ef6b5e
    • Sebastian Dröge's avatar
      ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are... · 31b67759
      Sebastian Dröge authored
      ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
      
      Original commit message from CVS:
      * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
      * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
      Add sane defaults for the 7 and 8 channel layouts as those are
      undefined in the Vorbis spec. Use NONE channel layouts when decoding
      more than 8 channels instead of erroring out. Fixes bug #535356.
      31b67759
  4. 28 May, 2008 5 commits
    • Wim Taymans's avatar
      Add theoraparse to the docs and fix some docs. · 1a3053b2
      Wim Taymans authored
      Original commit message from CVS:
      * docs/plugins/Makefile.am:
      * docs/plugins/gst-plugins-base-plugins-docs.sgml:
      * docs/plugins/gst-plugins-base-plugins-sections.txt:
      * ext/theora/theoraparse.c:
      Add theoraparse to the docs and fix some docs.
      1a3053b2
    • Wim Taymans's avatar
      gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition... · 2855fb48
      Wim Taymans authored
      gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
      
      Original commit message from CVS:
      * gst-libs/gst/cdda/gstcddabasesrc.c:
      (gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
      Fix EOS condition and track addition check, the track.end sector is
      included in the track. Fixes #533265.
      2855fb48
    • Mark Nauwelaerts's avatar
      gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT · 17b17a56
      Mark Nauwelaerts authored
      Original commit message from CVS:
      Patch by: Mark Nauwelaerts <manauw at skynet be>
      * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
      (gst_video_rate_flush_prev), (gst_video_rate_event),
      (gst_video_rate_chain):
      * gst/videorate/gstvideorate.h:
      React (more) to NEWSEGMENT
      Small adjustment in timestamp calculation to prevent mismatches
      Fixes #435633.
      17b17a56
    • Tim-Philipp Müller's avatar
      tests/examples/seek/seek.c: Initialise error to NULL as we should. · b82c4cee
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * tests/examples/seek/seek.c: (make_parselaunch_pipeline):
      Initialise error to NULL as we should.
      b82c4cee
    • Sebastian Dröge's avatar
      gst/adder/gstadder.c: Implement latency query. · 57c3aa9b
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/adder/gstadder.c: (gst_adder_query_duration),
      (gst_adder_query_latency), (gst_adder_query):
      Implement latency query.
      57c3aa9b
  5. 27 May, 2008 6 commits
    • Sebastian Dröge's avatar
      gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns · 4ccac97b
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/adder/gstadder.c: (gst_adder_query_duration):
      Correctly resync the iterator if gst_iterator_next() returns
      GST_ITERATOR_RESYNC.
      4ccac97b
    • Tim-Philipp Müller's avatar
      win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037). · 5d121dd6
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * win32/vs6/libgstpbutils.dsp:
      Add pbutils-enumtypes.c to sources (#518037).
      5d121dd6
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the... · 35e4b75b
      Wim Taymans authored
      gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
      (gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
      * gst-libs/gst/audio/gstaudioclock.h:
      Add method to inform the clock that the time starts from 0 again. We use
      this info to calculate a clock offset so that the time we report in
      internal_time is monotonically increasing, as required by the clock base
      class. Fixes #521761.
      API: GstAudioClock::gst_audio_clock_reset()
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_skew_slaving),
      (gst_base_audio_sink_change_state):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create), (gst_base_audio_src_change_state):
      Reset reported time when we (re)create the ringbuffer.
      35e4b75b
    • Tim-Philipp Müller's avatar
      ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally... · dc9eb0d6
      Tim-Philipp Müller authored
      ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
      
      Original commit message from CVS:
      * ext/alsa/gstalsamixertrack.c:
      (gst_alsa_mixer_track_update_alsa_capabilities):
      Make sure playback volumes aren't accidentally overwritten by
      capture volumes if an alsa mixer track has both playback and
      capture capabilities: we create two GstMixerTracks in that
      case, so make sure we query only the alsa capabilities that
      refer to the type of GstMixerTrack we created from the dual
      capability alsa element. Should fix issues with Audigy2 sound
      cards (#518082).
      dc9eb0d6
    • Tim-Philipp Müller's avatar
      tests/check/pipelines/oggmux.c: Don't use deprecated function. · 555feaa1
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * tests/check/pipelines/oggmux.c: (test_pipeline):
      Don't use deprecated function.
      555feaa1
    • Wim Taymans's avatar
      gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating... · 514b8fa4
      Wim Taymans authored
      gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
      
      Original commit message from CVS:
      * gst/playback/gstdecodebin2.c:
      (gst_decode_group_control_source_pad), (gst_decode_group_expose):
      Check for NULL cases and log them, creating ghostpads can, for example,
      fail when the pad returns wrong caps.
      * gst/playback/gstplaybin2.c: (perform_eos):
      When pushing out the EOS event, collect the return value and warn when
      something failed.
      514b8fa4
  6. 26 May, 2008 3 commits
  7. 25 May, 2008 1 commit
    • Tim-Philipp Müller's avatar
      Limit duration to a maximum of five seconds for tmplayer format where we can... · 206f9199
      Tim-Philipp Müller authored
      Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
      
      Original commit message from CVS:
      * gst/subparse/gstsubparse.c: (parser_state_init),
      (gst_sub_parse_format_autodetect), (handle_buffer):
      * gst/subparse/gstsubparse.h:
      * tests/check/elements/subparse.c: (test_tmplayer_style3b):
      Limit duration to a maximum of five seconds for tmplayer format where
      we can guess the duration only from the timestamp of the next line of
      text. We don't want to show a text for eternities just because nothing
      else is being said for a while.
      206f9199
  8. 23 May, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input... · 79a72514
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_chain),
      (gst_base_rtp_depayload_handle_sink_event),
      (gst_base_rtp_depayload_push_full),
      (gst_base_rtp_depayload_change_state):
      Check sequence numbers, mark input buffers with a discont flag for the
      subclass when we detected a gap, drop duplicate buffers. We do this
      because one can use the element without a jitterbuffer in front and we
      don't want to feed the subclasses invalid or reordered data.
      Do an error when the subclass did not provide a process function instead
      of crashing.
      Some other small cleanups.
      79a72514
  9. 22 May, 2008 4 commits
    • Tim-Philipp Müller's avatar
      gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here. · 747d52ad
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
      May just as well use the precalculated uvstride here.
      747d52ad
    • 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
  10. 21 May, 2008 12 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's avatar
      gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms. · 0f80e462
      Julien 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