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  1. Jun 18, 2008
    • Jan Schmidt's avatar
      Release 0.10.20 · 01e689e3
      Jan Schmidt authored
      Original commit message from CVS:
      Release 0.10.20
    • Jan Schmidt's avatar
      Update .po files · 7fd15d3d
      Jan Schmidt authored
      Original commit message from CVS:
      Update .po files
      7fd15d3d
    • Stefan Kost's avatar
      Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments. · 47335239
      Stefan Kost authored
      Original commit message from CVS:
      * docs/plugins/gst-plugins-bad-plugins-sections.txt:
      * examples/app/appsrc-ra.c:
      * examples/app/appsrc-seekable.c:
      * examples/app/appsrc-stream.c:
      * examples/app/appsrc-stream2.c:
      * ext/directfb/dfbvideosink.h:
      * ext/metadata/gstbasemetadata.c:
      * ext/metadata/gstbasemetadata.h:
      * ext/metadata/metadata.c:
      * ext/metadata/metadataexif.c:
      * ext/theora/theoradec.h:
      * gst/deinterlace2/gstdeinterlace2.h:
      * gst/deinterlace2/tvtime/speedy.c:
      * gst/deinterlace2/tvtime/speedy.h:
      * gst/deinterlace2/tvtime/vfir.c:
      Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
      comments.
      47335239
  2. Jun 16, 2008
    • Andy Wingo Wingo's avatar
      gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes) · 7ba91106
      Andy Wingo Wingo authored
      Original commit message from CVS:
      2008-06-16  Andy Wingo  <wingo@pobox.com>
      
      * gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
      (gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
      G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
      7ba91106
    • Stefan Kost's avatar
      Final round of doc updates. · 332fe998
      Stefan Kost authored
      Original commit message from CVS:
      * gst/rtpmanager/gstrtpjitterbuffer.c:
      * gst/speed/gstspeed.c:
      * gst/speexresample/gstspeexresample.c:
      * gst/videosignal/gstvideoanalyse.c:
      * gst/videosignal/gstvideodetect.c:
      * gst/videosignal/gstvideomark.c:
      * sys/dvb/gstdvbsrc.c:
      * sys/oss4/oss4-mixer.c:
      * sys/oss4/oss4-sink.c:
      * sys/oss4/oss4-source.c:
      * sys/wininet/gstwininetsrc.c:
      Final round of doc updates.
      332fe998
  3. Jun 13, 2008
    • Stefan Kost's avatar
      docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml · 4ad8ad3d
      Stefan Kost authored
      Original commit message from CVS:
      * docs/plugins/Makefile.am:
      * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
      * docs/plugins/gst-plugins-bad-plugins-sections.txt:
      * docs/plugins/gst-plugins-bad-plugins.args:
      * docs/plugins/gst-plugins-bad-plugins.hierarchy:
      * docs/plugins/gst-plugins-bad-plugins.interfaces:
      * docs/plugins/gst-plugins-bad-plugins.prerequisites:
      * docs/plugins/gst-plugins-bad-plugins.signals:
      * docs/plugins/inspect/plugin-alsaspdif.xml:
      * docs/plugins/inspect/plugin-amrwb.xml:
      * docs/plugins/inspect/plugin-app.xml:
      * docs/plugins/inspect/plugin-bayer.xml:
      * docs/plugins/inspect/plugin-bz2.xml:
      * docs/plugins/inspect/plugin-cdaudio.xml:
      * docs/plugins/inspect/plugin-cdxaparse.xml:
      * docs/plugins/inspect/plugin-dtsdec.xml:
      * docs/plugins/inspect/plugin-dvb.xml:
      * docs/plugins/inspect/plugin-dvdspu.xml:
      * docs/plugins/inspect/plugin-faac.xml:
      * docs/plugins/inspect/plugin-faad.xml:
      * docs/plugins/inspect/plugin-fbdevsink.xml:
      * docs/plugins/inspect/plugin-festival.xml:
      * docs/plugins/inspect/plugin-filter.xml:
      * docs/plugins/inspect/plugin-flvdemux.xml:
      * docs/plugins/inspect/plugin-freeze.xml:
      * docs/plugins/inspect/plugin-gsm.xml:
      * docs/plugins/inspect/plugin-gstinterlace.xml:
      * docs/plugins/inspect/plugin-gstrtpmanager.xml:
      * docs/plugins/inspect/plugin-h264parse.xml:
      * docs/plugins/inspect/plugin-interleave.xml:
      * docs/plugins/inspect/plugin-jack.xml:
      * docs/plugins/inspect/plugin-ladspa.xml:
      * docs/plugins/inspect/plugin-metadata.xml:
      * docs/plugins/inspect/plugin-mms.xml:
      * docs/plugins/inspect/plugin-modplug.xml:
      * docs/plugins/inspect/plugin-mpeg2enc.xml:
      * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
      * docs/plugins/inspect/plugin-mpegtsparse.xml:
      * docs/plugins/inspect/plugin-mpegvideoparse.xml:
      * docs/plugins/inspect/plugin-musepack.xml:
      * docs/plugins/inspect/plugin-musicbrainz.xml:
      * docs/plugins/inspect/plugin-mve.xml:
      * docs/plugins/inspect/plugin-mythtv.xml
      * docs/plugins/inspect/plugin-nas.xml:
      * docs/plugins/inspect/plugin-neon.xml:
      * docs/plugins/inspect/plugin-nsfdec.xml:
      * docs/plugins/inspect/plugin-nuvdemux.xml:
      * docs/plugins/inspect/plugin-oss4.xml
      * docs/plugins/inspect/plugin-rawparse.xml:
      * docs/plugins/inspect/plugin-real.xml:
      * docs/plugins/inspect/plugin-replaygain.xml:
      * docs/plugins/inspect/plugin-rfbsrc.xml:
      * docs/plugins/inspect/plugin-sdl.xml:
      * docs/plugins/inspect/plugin-sdp.xml:
      * docs/plugins/inspect/plugin-selector.xml:
      * docs/plugins/inspect/plugin-sndfile.xml:
      * docs/plugins/inspect/plugin-soundtouch.xml:
      * docs/plugins/inspect/plugin-spcdec.xml:
      * docs/plugins/inspect/plugin-speed.xml:
      * docs/plugins/inspect/plugin-speexresample.xml:
      * docs/plugins/inspect/plugin-stereo.xml:
      * docs/plugins/inspect/plugin-subenc.xml
      * docs/plugins/inspect/plugin-timidity.xml:
      * docs/plugins/inspect/plugin-tta.xml:
      * docs/plugins/inspect/plugin-vcdsrc.xml:
      * docs/plugins/inspect/plugin-videosignal.xml:
      * docs/plugins/inspect/plugin-vmnc.xml:
      * docs/plugins/inspect/plugin-wildmidi.xml:
      * docs/plugins/inspect/plugin-x264.xml:
      * docs/plugins/inspect/plugin-xvid.xml:
      * docs/plugins/inspect/plugin-y4menc.xml:
      * ext/amrwb/gstamrwbdec.c:
      * ext/amrwb/gstamrwbenc.c:
      * ext/amrwb/gstamrwbparse.c:
      * ext/dc1394/gstdc1394.c:
      * ext/directfb/dfbvideosink.c:
      * ext/ivorbis/vorbisdec.c:
      * ext/jack/gstjackaudiosink.c:
      * ext/mpeg2enc/gstmpeg2enc.cc:
      * ext/mplex/gstmplex.cc:
      * ext/musicbrainz/gsttrm.c:
      * ext/mythtv/gstmythtvsrc.c:
      * ext/theora/theoradec.c:
      * ext/timidity/gsttimidity.c:
      * ext/timidity/gstwildmidi.c:
      * gst-libs/gst/app/gstappsink.c:
      * gst/deinterlace/gstdeinterlace.c:
      * gst/dvdspu/gstdvdspu.c:
      * gst/festival/gstfestival.c:
      * gst/freeze/gstfreeze.c:
      * gst/interleave/deinterleave.c:
      * gst/interleave/interleave.c:
      * gst/modplug/gstmodplug.cc:
      * gst/nuvdemux/gstnuvdemux.c:
      Add missing elements to docs. Fix doc-markup: use convinience syntax
      for examples (produces valid docbook), add several refsec2 when we
      have several titles. Fix some types.
      4ad8ad3d
  4. Jun 12, 2008
    • Wim Taymans's avatar
      examples/app/: Add beefed up example app from bug #413418. It now also uses... · c30d4797
      Wim Taymans authored
      examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
      
      Original commit message from CVS:
      * examples/app/.cvsignore:
      * examples/app/Makefile.am:
      * examples/app/appsink-src.c: (on_new_buffer_from_source),
      (on_source_message), (on_sink_message), (main):
      Add beefed up example app from bug #413418. It now also uses appsink
      instead of fakesink for more ultimate coolness.
      * 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_create),
      (gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
      (gst_app_src_end_of_stream):
      * gst-libs/gst/app/gstappsrc.h:
      Add block property to allow push based implementation to block when we
      fill up the appsrc queues.
      Emit the enough-data signal while releasing our lock.
      c30d4797
    • Stefan Kost's avatar
      examples/app/.cvsignore: Ignore more. · 0375b4a6
      Stefan Kost authored
      Original commit message from CVS:
      * examples/app/.cvsignore:
      Ignore more.
      0375b4a6
    • Stefan Kost's avatar
      Do not use short_description in section docs for elements. We extract them... · e54b324d
      Stefan Kost authored
      Do not use short_description in section docs for elements. We extract them from element details and there will be war...
      
      Original commit message from CVS:
      * ext/dc1394/gstdc1394.c:
      * ext/ivorbis/vorbisdec.c:
      * ext/jack/gstjackaudiosink.c:
      * ext/metadata/gstmetadatademux.c:
      * ext/mythtv/gstmythtvsrc.c:
      * ext/theora/theoradec.c:
      * gst-libs/gst/app/gstappsink.c:
      * gst/bayer/gstbayer2rgb.c:
      * gst/deinterlace/gstdeinterlace.c:
      * gst/rawparse/gstaudioparse.c:
      * gst/rawparse/gstvideoparse.c:
      * gst/rtpmanager/gstrtpbin.c:
      * gst/rtpmanager/gstrtpclient.c:
      * gst/rtpmanager/gstrtpjitterbuffer.c:
      * gst/rtpmanager/gstrtpptdemux.c:
      * gst/rtpmanager/gstrtpsession.c:
      * gst/rtpmanager/gstrtpssrcdemux.c:
      * gst/selector/gstinputselector.c:
      * gst/selector/gstoutputselector.c:
      * gst/videosignal/gstvideoanalyse.c:
      * gst/videosignal/gstvideodetect.c:
      * gst/videosignal/gstvideomark.c:
      * sys/oss4/oss4-mixer.c:
      * sys/oss4/oss4-sink.c:
      * sys/oss4/oss4-source.c:
      Do not use short_description in section docs for elements. We extract
      them from element details and there will be warnings if they differ.
      Also fixing up the ChangeLog order.
      e54b324d
  5. Jun 11, 2008
    • Jan Schmidt's avatar
      configure.ac: 0.10.19.3 pre-release · 5dd552cf
      Jan Schmidt authored
      Original commit message from CVS:
      * configure.ac:
      0.10.19.3 pre-release
      5dd552cf
    • David Schleef's avatar
      gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32. · 526b2e63
      David Schleef authored
      Original commit message from CVS:
      * gst-libs/gst/rtsp/gstrtspconnection.c:
      Fix build on win32.
      Patch By: David Schleef <ds@schleef.org>
      Fixes: #536874
      526b2e63
    • Sebastian Dröge's avatar
      ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if... · bb595d8f
      Sebastian Dröge authored
      ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
      
      Original commit message from CVS:
      * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
      (gst_gio_base_src_create):
      * ext/gio/gstgiobasesrc.h:
      Try to read the requested number of bytes, even if the first
      read returns less than requested, until nothing is read anymore
      or we have the requested amount of bytes. This fixes playback of
      files via Samba as Samba only allows to read 64k at once.
      Implement a caching algorithm that makes sure that we read at
      least 4k of data every time. Some elements will try to read a few
      bytes, then seek, read again a few bytes and so on and this is
      painfully slow as every operation has to go over DBus if GVfs is
      used as backend.
      Fixes bug #536849 and #536848.
      * ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
      (gst_gio_src_check_get_range):
      Override check_get_range() to blacklist http/https URIs
      and whitelist file URIs. More to be added on demand.
      bb595d8f
  6. Jun 06, 2008
    • Wim Taymans's avatar
      examples/app/: Added 3 more example application for using appsrc in... · 593d4b1a
      Wim Taymans authored
      examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
      
      Original commit message from CVS:
      * examples/app/Makefile.am:
      * examples/app/appsrc-ra.c: (feed_data), (seek_data),
      (found_source), (bus_message), (main):
      * examples/app/appsrc-seekable.c: (feed_data), (seek_data),
      (found_source), (bus_message), (main):
      * examples/app/appsrc-stream2.c: (feed_data), (found_source),
      (bus_message), (main):
      Added 3 more example application for using appsrc in random-access mode,
      pull-mode streaming and pull mode seekable.
      * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
      (gst_app_src_start), (gst_app_src_do_get_size),
      (gst_app_src_create):
      * gst-libs/gst/app/gstappsrc.h:
      Make stream-type property writable.
      Unset flushing when starting so that we reuse appsrc.
      Inform basesrc about the configured size.
      Emit seek-data signal when we are going to a different offset in
      random-access mode.
      593d4b1a
    • Wim Taymans's avatar
      examples/app/appsrc-stream.c: Use deep-notify until we can depend on a... · 1cb26cc0
      Wim Taymans authored
      examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
      
      Original commit message from CVS:
      * examples/app/appsrc-stream.c: (found_source), (main):
      Use deep-notify until we can depend on a playbin2 with support for the
      source property.
      1cb26cc0
  7. Jun 05, 2008
    • Wim Taymans's avatar
      examples/app/: Added an example on how to use appsrc in playbin in streaming... · 20d64607
      Wim Taymans authored
      examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
      
      Original commit message from CVS:
      * examples/app/.cvsignore:
      * examples/app/Makefile.am:
      * examples/app/appsrc-stream.c: (read_data), (start_feed),
      (stop_feed), (found_source), (bus_message), (main):
      Added an example on how to use appsrc in playbin in streaming mode from
      an mmapped file.
      * examples/app/appsrc_ex.c: (main):
      Set pipeline to NULL to free queued buffers.
      * gst-libs/gst/app/gstapp-marshal.list:
      * gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
      (gst_app_src_class_init), (gst_app_src_init),
      (gst_app_src_flush_queued), (gst_app_src_dispose),
      (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_is_seekable),
      (gst_app_src_check_get_range), (gst_app_src_do_seek),
      (gst_app_src_create), (gst_app_src_set_stream_type),
      (gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
      (gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
      (gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
      (gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
      (gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
      * gst-libs/gst/app/gstappsrc.h:
      Measure max queue size in bytes instead.
      Add support for 3 modes of operation, streaming, seekable and
      random-access, making basesrc handle the scheduling modes for each.
      Add appsrc:// uri handler so that automatic plugging can be done from
      playbin2 or uridecodebin, for example.
      Added support for custom segment formats.
      Add support for push and pull based operations from the application.
      Expand the methods so that errors can be detected.
      Flush the queued buffers on seeks and when shutting down.
      Add signals to inform the app that a seek must happen.
      20d64607
    • Jan Schmidt's avatar
      configure.ac: 0.10.19.2 pre-release · 71546edc
      Jan Schmidt authored
      Original commit message from CVS:
      * configure.ac:
      0.10.19.2 pre-release
      71546edc
  8. Jun 04, 2008
    • Jan Schmidt's avatar
      win32/common/: Add new API functions to the dll exports · cb8b68c5
      Jan Schmidt authored
      Original commit message from CVS:
      * win32/common/libgstrtsp.def:
      * win32/common/libgsttag.def:
      Add new API functions to the dll exports
      cb8b68c5
    • Michael Smith's avatar
      gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created... · 2fdd607e
      Michael Smith authored
      gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
      
      Original commit message from CVS:
      * gst/playback/gstplaybasebin.c:
      Disconnect signals from decodebins we created before we remove it from
      playbin, to avoid crashes if the decodebin is eventually disposed after
      the playbin itself (possible if the app takes a reference on the
      decodebin).
      Fixes #536521.
      2fdd607e
    • Tim-Philipp Müller's avatar
      gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use... · 93db55c0
      Tim-Philipp Müller authored
      gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
      
      Original commit message from CVS:
      * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
      (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
      (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
      (h264_video_type_find), (mpeg_video_stream_type_find),
      (dv_type_find), (mmsh_type_find):
      Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
      copy caps for no good reason (this may be desirable to make it easier
      to detect leaks, but then it should probably be done for all caps
      in the typefinder somewhere).
      93db55c0
    • Peter Kjellerstedt's avatar
      tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built. · ec07ea99
      Peter Kjellerstedt authored
      Original commit message from CVS:
      * tests/check/Makefile.am:
      Do not try to run the check tests for subparse unless it has been
      built.
      ec07ea99
    • Peter Kjellerstedt's avatar
      tests/check/pipelines/streamheader.c: Do not try to run a test which requires... · 4d05d8ab
      Peter Kjellerstedt authored
      tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
      
      Original commit message from CVS:
      * tests/check/pipelines/streamheader.c: (buffer_probe_cb),
      (test_multifdsink_gdp_vorbisenc), (streamheader_suite):
      Do not try to run a test which requires vorbisenc unless we have
      actually built it.
      4d05d8ab
    • Peter Kjellerstedt's avatar
      gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards. · 26cd5ea1
      Peter Kjellerstedt authored
      Original commit message from CVS:
      * gst-libs/gst/rtsp/gstrtspconnection.c:
      (gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
      (gst_rtsp_connection_clear_auth_params),
      (gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
      * gst-libs/gst/rtsp/gstrtspconnection.h:
      Add a couple of missing argument guards.
      Add a way of setting the DSCP for an RTSP connection.
      Add an accessor method for the ip member of GstRTSPConnection as all
      members are supposed to be private.
      26cd5ea1
    • Peter Kjellerstedt's avatar
      gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses. · c1405283
      Peter Kjellerstedt authored
      Original commit message from CVS:
      * gst/tcp/gstmultifdsink.c: (setup_dscp_client):
      Fixed accidental use of IPv4 options for all IPv6 addresses.
      c1405283
    • Tim-Philipp Müller's avatar
      gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags. · 44e087f8
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * gst-libs/gst/interfaces/mixertrack.h:
      Document mixer track flags.
      44e087f8
    • Antoine Tremblay's avatar
      gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the... · be2f6a80
      Antoine Tremblay authored and Sebastian Dröge's avatar Sebastian Dröge committed
      gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
      
      Original commit message from CVS:
      Patch by: Antoine Tremblay <hexa00 at gmail dot com>
      * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
      Don't set caps on the buffers that contain a copy of the buffer
      including the caps of them resulting in an always increasing refcount
      of the caps and insanely large caps. Instead include a buffer without
      caps in the new caps. Fixes bug #536475.
      be2f6a80
    • Sebastian Dröge's avatar
      gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct... · d57ab7cf
      Sebastian Dröge authored
      gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
      
      Original commit message from CVS:
      * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
      Transform a given PAR to a range on the struct with the generic
      height/width instead of the struct with the possibly restricted
      height/width.
      d57ab7cf
    • Sebastian Dröge's avatar
      gst/videoscale/gstvideoscale.c: Prefer the given format if it contains... · 8b14d081
      Sebastian Dröge authored
      gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
      
      Original commit message from CVS:
      * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
      Prefer the given format if it contains something stricter than [1,MAX]
      for height or width and only put a structure that requires rescaling
      as second. This makes it possible to use videoscale in pipelines where
      the source can actually produce the wanted height/width but usually
      selects a different one from the requested.
      8b14d081
  9. Jun 03, 2008
  10. Jun 02, 2008
  11. May 31, 2008
    • Mark Nauwelaerts's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new... · 9fa61c52
      Mark Nauwelaerts authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init):
      Add a gtk-doc chunk for the new properties to have a Since: indication.
      9fa61c52
    • Mark Nauwelaerts's avatar
      ChangeLog surgery, mark API change · 1985500e
      Mark Nauwelaerts authored
      Original commit message from CVS:
      ChangeLog surgery, mark API change
      1985500e
    • 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
  12. May 30, 2008
    • 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
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