1. 19 May, 2008 2 commits
    • Tim-Philipp Müller's avatar
      gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder... · 7cb1276d
      Tim-Philipp Müller authored
      gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
      
      Original commit message from CVS:
      * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
      Use data scan helper in aac typefinder and stop scanning
      for headers when we've found a type. Also fix potential invalid
      memory access when calculating the frame length.
      7cb1276d
    • Tim-Philipp Müller's avatar
      gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return... · cfc8f3c0
      Tim-Philipp Müller authored
      gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
      
      Original commit message from CVS:
      * gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
      (mpeg_sys_is_valid_pack):
      Don't modify scan context when we return FALSE in ensure_data, so
      it's possible to continue scanning, and we don't end up with a NULL
      data pointer and a positive size, which might bite us the next time
      we're called. Small constification.
      cfc8f3c0
  2. 16 May, 2008 1 commit
  3. 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
  4. 13 May, 2008 6 commits
    • hansenlabs's avatar
      gst/tcp/: Fix regression in clientsrc because we did not add the fd to the... · b9bc12af
      hansenlabs authored
      gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
      
      Original commit message from CVS:
      Patch by: Hannes Bistry <hannesb at gmx dot de>
      * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
      * gst/tcp/gsttcpserversink.c:
      (gst_tcp_server_sink_handle_server_read),
      (gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
      Fix regression in clientsrc because we did not add the fd to the poll
      set anymore. Fixes #532364.
      Do some cleanups here and there.
      b9bc12af
    • Sebastian Dröge's avatar
      gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass. · 05349cc3
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
      * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
      * gst/playback/gstplay-marshal.list:
      * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
      Use correct marshallers. GstCaps are a boxed type and no GObject
      subclass.
      05349cc3
    • Sebastian Dröge's avatar
      win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param()... · 5800b1ac
      Sebastian Dröge authored
      win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.
      
      Original commit message from CVS:
      * win32/common/libgstrtsp.def:
      Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
      symbols.
      5800b1ac
    • Sjoerd Simons's avatar
      tests/check/elements/audioresample.c: Add unit test for the latest... · fd84ec0c
      Sjoerd Simons authored
      tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
      
      Original commit message from CVS:
      Patch by: Sjoerd Simons <sjoerd at luon dot net>
      * tests/check/elements/audioresample.c:
      (live_switch_alloc_only_48000), (live_switch_get_sink_caps),
      (live_switch_push), (GST_START_TEST):
      Add unit test for the latest basetransform negotiation changes.
      See bug #526768.
      fd84ec0c
    • Sebastian Dröge's avatar
      gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width. · 4d587084
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
      Fix nv12<->nv21 conversion if stride is larger than width.
      4d587084
    • j^'s avatar
      ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use... · 1a154e1d
      j^ authored
      ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...
      
      Original commit message from CVS:
      Patch by: j^ <j at oil21 dot org>
      * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
      (gst_ogg_pad_parse_skeleton_fisbone):
      * ext/ogg/gstoggdemux.h:
      Parse presentation time from skeleton streams and use it as offset
      for the timestamps. Fixes bug #530068.
      1a154e1d
  5. 12 May, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to... · 0c9b1398
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
      Revert previous patch that attempted to more accurately calculate the
      initial offset between master and slave clock. The best thing we can do
      in general is take the time of both clocks as the diff since we don't
      know when the actual preroll happened.
      0c9b1398
  6. 11 May, 2008 1 commit
  7. 10 May, 2008 2 commits
    • Tim-Philipp Müller's avatar
      gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use... · fed34307
      Tim-Philipp Müller authored
      gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
      
      Original commit message from CVS:
      * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
      Don't do lots of 4-byte peeks, but use the 'new' data scan helper
      for this instead; don't check if we've found enough markers after
      each and every step, it's enough to do that only if we've actually
      found a new marker.
      Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
      fed34307
    • Tim-Philipp Müller's avatar
      gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning... · 104fed4d
      Tim-Philipp Müller authored
      gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
      
      Original commit message from CVS:
      * gst/typefind/gsttypefindfunctions.c:
      (DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
      (data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
      (mpeg_video_stream_type_find):
      Move scan helper thingy to the beginning of the file so we can use
      it in other typefind functions. Rename it to something more
      generic. Also improve handling of things towards the end of the
      typefind data: peek as much as we can if we know the size of the
      data, rather than just min_size.
      104fed4d
  8. 09 May, 2008 3 commits
    • Jan Schmidt's avatar
      Document the GstTuner and GstColorBalance interfaces, and some other random... · f11cf32c
      Jan Schmidt authored
      Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
      
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * gst-libs/gst/interfaces/colorbalance.c:
      * gst-libs/gst/interfaces/colorbalance.h:
      * gst-libs/gst/interfaces/colorbalancechannel.c:
      * gst-libs/gst/interfaces/colorbalancechannel.h:
      * gst-libs/gst/interfaces/tuner.c:
      * gst-libs/gst/interfaces/tunerchannel.c:
      * gst-libs/gst/interfaces/tunerchannel.h:
      * gst-libs/gst/interfaces/tunernorm.c:
      * gst-libs/gst/interfaces/tunernorm.h:
      * gst-libs/gst/video/video.c:
      * gst-libs/gst/video/video.h:
      Document the GstTuner and GstColorBalance interfaces, and some
      other random API functions that needed it. 70% symbol coverage, woo.
      f11cf32c
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but... · fc523e04
      Wim Taymans authored
      gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
      Choose to allocate one less segment but require one additional segment
      as latency.
      * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
      No need to increment the number of segments in the source.
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_get_time), (clock_convert_external),
      (gst_base_audio_sink_resample_slaving),
      (gst_base_audio_sink_skew_slaving),
      (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
      (gst_base_audio_sink_async_play):
      Remove adding latency when returning the internal time while subtracting
      it again when we use the value a little later.
      When calculating the end timestamp, we are making a rounding error
      with the current algorithm. Ensure that we don't accumulate these
      rounding errors when aligning samples by not resampling at all if we
      don't need to. Fixes #419351.
      Make the initial calibration of the clock slaving a little more
      predictable and accurate. Also handle the case where we don't do
      clock slaving.
      fc523e04
    • Sebastian Dröge's avatar
      gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions... · 531c6fb4
      Sebastian Dröge authored
      gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
      
      Original commit message from CVS:
      Based on a patch by:
      Björn Benderius <bjoern dot benderius at axis dot com>
      * gst/ffmpegcolorspace/avcodec.h:
      * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
      (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
      (gst_ffmpegcsp_avpicture_fill):
      * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
      * gst/ffmpegcolorspace/imgconvert_template.h:
      Add conversions from/to NV12 and NV21 and conversions between those
      two formats. Fixes bug #532166.
      531c6fb4
  9. 08 May, 2008 3 commits
    • Edward Hervey's avatar
      gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as... · 9fa3d7a2
      Edward Hervey authored
      gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
      
      Original commit message from CVS:
      * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
      Abort the h264 typefinding as soon as _peek() doesn't return anything,
      which happens for example with files smaller than 128kb.
      9fa3d7a2
    • Wouter Cloetens's avatar
      gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065. · a8a2b9c7
      Wouter Cloetens authored
      Original commit message from CVS:
      Patch by: Wouter Cloetens <zombie at e2big dot org>
      * gst-libs/gst/rtsp/Makefile.am:
      * gst-libs/gst/rtsp/gstrtspconnection.c:
      (gst_rtsp_connection_create), (md5_digest_to_hex_string),
      (auth_digest_compute_hex_urp), (auth_digest_compute_response),
      (add_auth_header), (gst_rtsp_connection_free),
      (gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
      (gst_rtsp_connection_set_auth_param),
      (gst_rtsp_connection_clear_auth_params):
      * gst-libs/gst/rtsp/gstrtspconnection.h:
      Add Digest authorization support for RTSP connections. See #532065.
      * gst-libs/gst/rtsp/md5.c:
      * gst-libs/gst/rtsp/md5.h:
      Yeap, another md5 implementation until we can depend on a glib that has
      support for it.
      a8a2b9c7
    • Sjoerd Simons's avatar
      gst/audioresample/gstaudioresample.c: Let audioresample use the buffer... · 09163ca3
      Sjoerd Simons authored
      gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
      
      Original commit message from CVS:
      Patch by: Sjoerd Simons <sjoerd at luon dot net>
      * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
      Let audioresample use the buffer allocation of basetransform instead
      of it's own stuff.
      * tests/check/elements/audioresample.c: (alloc_only_48000),
      (GST_START_TEST), (audioresample_suite):
      Add unit test for the recent basetransform bugfix, where upstream
      changes caps to something that can't be passed through anymore.
      09163ca3
  10. 07 May, 2008 3 commits
    • Ole Andre Vadla Ravnaas's avatar
      win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC... · 7a22e13f
      Ole Andre Vadla Ravnaas authored
      win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
      
      Original commit message from CVS:
      * win32/common/config.h.in:
      Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
      use the real thing than having "???" unconditionally.
      7a22e13f
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter. · 09f7dee8
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_query):
      Report the latency with the new seglatency parameter.
      * gst-libs/gst/audio/gstringbuffer.c:
      (gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
      (gst_ring_buffer_acquire):
      * gst-libs/gst/audio/gstringbuffer.h:
      Add new field to the ringbufferspec to specify the expected latency
      between the underlying device read/write pointer, this is needed
      when writing sinks that sit a little closer to the hardware.
      Add some more docs for other fields.
      09f7dee8
    • Wim Taymans's avatar
      gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore. · 8c3b00be
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/app/.cvsignore:
      * gst-libs/gst/app/Makefile.am:
      * gst-libs/gst/app/gstapp-marshal.list:
      Add marshal.list, make it compile and add to cvsignore.
      * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
      (gst_app_sink_stop):
      Small cleanups.
      * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
      (gst_app_src_init), (gst_app_src_set_property),
      (gst_app_src_get_property), (gst_app_src_unlock),
      (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
      (gst_app_src_create), (gst_app_src_set_caps),
      (gst_app_src_get_caps), (gst_app_src_set_size),
      (gst_app_src_get_size), (gst_app_src_set_seekable),
      (gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
      (gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
      (gst_app_src_end_of_stream):
      * gst-libs/gst/app/gstappsrc.h:
      Beat appsrc in shape, add signals and actions.
      Add some docs.
      Add properties for caps, size, seekability and max-buffers.
      Fix unlock/stop code.
      8c3b00be
  11. 06 May, 2008 4 commits
    • Sebastian Dröge's avatar
      gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process... · b9a28502
      Sebastian Dröge authored
      gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
      
      Original commit message from CVS:
      * gst/volume/gstvolume.c: (volume_transform_ip):
      Return NOT_NEGOTIATED if we didn't set a process function yet for some
      reason instead of crashing later. Might fix bug #509125.
      b9a28502
    • Tim-Philipp Müller's avatar
      gst/audioconvert/: Add support for more than 8 channels and NONE channel... · fd54092a
      Tim-Philipp Müller authored
      gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
      
      Original commit message from CVS:
      Based on a patch by: Tim-Philipp Müller  <tim.muller at collabora co uk>
      * gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
      * gst/audioconvert/audioconvert.h:
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_parse_caps),
      (structure_has_fixed_channel_positions),
      (gst_audio_convert_transform_caps):
      * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
      Add support for more than 8 channels and NONE channel layouts. For
      more than 8 channels no channel conversion is supported yet, only
      format conversions are supported. Fixes bug #398033.
      * tests/check/elements/audioconvert.c: (verify_convert),
      (GST_START_TEST), (audioconvert_suite):
      Add some unit tests by Tim for checking the NONE channel layouts
      and more than 8 channels and add some more unit tests for channel
      conversions.
      fd54092a
    • Wim Taymans's avatar
      gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to... · 4a3db41f
      Wim Taymans authored
      gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
      
      Original commit message from CVS:
      * gst/playback/gstdecodebin2.c: (connect_pad):
      When autoplugging fails, set the element back to NULL before
      unreffing it.
      4a3db41f
    • Sebastian Dröge's avatar
      win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols. · 9854bd07
      Sebastian Dröge authored
      Original commit message from CVS:
      * win32/common/libgstaudio.def:
      Add gst_base_audio_src_[sg]et_slave_method() to the exported
      symbols.
      9854bd07
  12. 05 May, 2008 5 commits
    • Sebastian Dröge's avatar
      gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces. · 9333eb48
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/subparse/samiparse.c: (handle_start_sync),
      (end_sami_element), (characters_sami):
      Remove trailing, leading and double whitespaces.
      Correctly timestamp buffers and output the last buffer too.
      * tests/check/elements/subparse.c: (GST_START_TEST),
      (subparse_suite):
      Add a simple unit test for SAMI parsing.
      9333eb48
    • Young-Ho Cha's avatar
      gst/subparse/samiparse.c: Only output characters inside the "sync" elements.... · 76e3ffb6
      Young-Ho Cha authored
      gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
      
      Original commit message from CVS:
      Patch by: Young-Ho Cha <ganadist at chollian dot net>
      * gst/subparse/samiparse.c: (handle_start_sync),
      (start_sami_element), (end_sami_element), (characters_sami),
      (sami_context_reset):
      Only output characters inside the "sync" elements. There could be
      other elements like "style" that have some content but should
      not be printed. Fixes bug #467911.
      76e3ffb6
    • Wim Taymans's avatar
      gst-libs/gst/app/gstappsink.*: Start some docs. · 1275eda1
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
      (gst_app_sink_init), (gst_app_sink_set_property),
      (gst_app_sink_get_property), (gst_app_sink_unlock_start),
      (gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
      (gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
      (gst_app_sink_preroll), (gst_app_sink_render),
      (gst_app_sink_set_caps), (gst_app_sink_set_drop),
      (gst_app_sink_get_drop):
      * gst-libs/gst/app/gstappsink.h:
      Start some docs.
      Add property to drop buffers when the queue is filled
      Fix unlocking and flushing when the queues are filled.
      1275eda1
    • Sebastian Dröge's avatar
      gst/playback/: Allow setting -1 as current-audio to mute the current audio... · de277a5b
      Sebastian Dröge authored
      gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
      
      Original commit message from CVS:
      * gst/playback/gstplaybasebin.c: (set_audio_mute),
      (set_active_source):
      * gst/playback/gstplaybasebin.h:
      * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
      (playbin_set_audio_mute):
      Allow setting -1 as current-audio to mute the current audio stream,
      similar to what is done for subtitles. Fixes bug #342294.
      de277a5b
    • Edward Hervey's avatar
      gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn. · b98072f9
      Edward Hervey authored
      Original commit message from CVS:
      * gst-libs/gst/pbutils/descriptions.c: (formats):
      It's SorensOn and not SorensEn.
      b98072f9
  13. 04 May, 2008 2 commits
    • Tim-Philipp Müller's avatar
      gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video. · 6451cbf5
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * gst-libs/gst/pbutils/descriptions.c: (formats):
      Fix description of video/x-flash-video.
      6451cbf5
    • Sebastian Dröge's avatar
      Remove some unused code. · 83f07293
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
      * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
      * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
      * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
      Remove some unused code.
      * gst/audioconvert/gstaudioquantize.c:
      (gst_audio_quantize_free_noise_shaping):
      Don't return before freeing the noise shaping history.
      83f07293
  14. 03 May, 2008 2 commits
    • Tim-Philipp Müller's avatar
      tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962. · 1157de77
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * tests/check/elements/subparse.c: (do_test),
      (test_tmplayer_style3b), (subparse_suite):
      Add unit test for the tmplayer variant from bug #530962.
      1157de77
    • Tim-Philipp Müller's avatar
      gst/subparse/: Fix parsing of tmplayer subtitle variant where every single... · 005c1c86
      Tim-Philipp Müller authored
      gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
      
      Original commit message from CVS:
      * gst/subparse/gstsubparse.c: (handle_buffer),
      (gst_sub_parse_sink_event):
      * gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
      (tmplayer_parse_line):
      Fix parsing of tmplayer subtitle variant where every single line contains
      text and there isn't an empty line after each line to determine the
      duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
      making sure that we push out the last line of text without a duration if
      there's still text left in the buffer at the end.
      005c1c86