gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...

Original commit message from CVS:
* gst/subparse/samiparse.c: (sami_context_push_state),
(sami_context_pop_state), (start_sami_element), (end_sami_element):
Some versions of libxml seem to be very picky as to strict formatting
of the input and never 'close' the final </body> tag.
In order to fix that bad behaviour, we trigger the flushing of
remaining data on both </body> and </sami>.
Fixes #557365

gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...

Original commit message from CVS:
Patch by: Guillaume Emont <guillaume at fluendo dot com>
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinders for MS Word files and OS X .DS_Store files to
prevent them to be recognized as MPEG files. Fixes bug #564098.

Original commit message from CVS:
* gst/playback/gstplaysink.c: (gen_audio_chain),
(gst_play_sink_reconfigure):
Add some more debug info.
Fix linking of just an encoded sink.
Handle failure to create a sink chain more gracefully than crashing.

Original commit message from CVS:
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
Pushing 10 buffers is enough to run the test.

Original commit message from CVS:
* tests/examples/seek/seek.c: (do_seek), (stop_cb),
(skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
(main):
Hook up the SKIP seek flag.

gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.

Original commit message from CVS:
* gst/playback/gstplaybin2.c: (pad_added_cb):
Error out with a missing-plugin error when the input-selector was not
found.
* gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
Indentation.

Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
(gst_play_sink_send_event), (gst_play_sink_change_state):
Use G_DEFINE_TYPE.
Try to set the selected sink to READY before using it. This will allow
for detection of incompatible formats sooner.
Don't cause a fatal error when conversion elements are missing but post
a missing-element message and a warning instead because things might
still link and run fine.
Simplyfy the construction of audio and video sink chains.

Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
(gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
init from glib.

Original commit message from CVS:
Patch by: Luis Menina <liberforce at freeside dot fr>
* gst-libs/gst/floatcast/floatcast.h:
* gst/typefind/gsttypefindfunctions.c:
Include glib.h instead of a specific GLib header. Including single
GLib headers is deprecated. Fixes bug #563904.

gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).

Original commit message from CVS:
2008-12-09  Julien Moutte  <julien@fluendo.com>

* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Support higher max audio rates for some formats (WAV, Vorbis, LPCM).

Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
Fix handling of odd chunks in riff metadata.

gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...

Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_before_transform), (volume_transform_ip):
Use new basetransform vmethod to reconfigure the dynamic properties and
any pending volume/mute changes. Fixes #563508.

configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...

Original commit message from CVS:
* configure.ac:
First check for "theoraenc theoradec" and if that failed check
for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
deprecate the latter. Also linking on Windows fails with just "theora"
and the version check would fail for the release candidates.
Fixes bug #563718.

Original commit message from CVS:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
Add basic docs to decodebin and link to decodebin from decodebin2.

Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement gst_rtcp_packet_remove(). Fixes #563174.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add unit test for some RTCP functions.

configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.

Original commit message from CVS:
* configure.ac:
Apparently AC_CONFIG_MACRO_DIR breaks when using more
than one macro directory, reverting last change.

Original commit message from CVS:
* configure.ac:
Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
our M4 macros.

Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
Clear all flags on buffers returned from the image pool.
Fixes #563143

gst-libs/gst/app/gstappsrc.c: Don't forget to release the lock again if we bail out because some pad is flushing or w...

Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
Don't forget to release the lock again if we bail out because some
pad is flushing or we've reached EOS, otherwise things will lock up
next time _push_buffer() is called (#562802).

Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...

Original commit message from CVS:
Patch by: Cygwin Ports maintainer
<yselkowitz at users dot sourceforge dot net>
* autogen.sh:
* configure.ac:
Require gettext 0.17 because older versions don't mix with libtool
2.2. At build time an older gettext version will still work.
Fixes bug #556091.

Original commit message from CVS:
fix build

Original commit message from CVS:
* 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-videorate.xml:
* gst/speexresample/gstspeexresample.c:
Update documentation of speexresample for the new element name.

gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.

Original commit message from CVS:
* gst/speexresample/README:
Update README with the latest diff between the Speex resampler
and our copy.

gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.

Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (plugin_init):
Update the debug category from speex_resample to audioresample.

Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
* tests/check/elements/audioresample.c:
Remove audioresample files.

Original commit message from CVS:
* docs/plugins/inspect/plugin-audioresample.xml:
Regenerated for library filename change.

Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...

Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
* 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:
* gst/speexresample/gstspeexresample.c: (plugin_init):
* gst/speexresample/Makefile.am:
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(GST_START_TEST), (test_pipeline):
Rename the moved speexresample to audioresample, integrate into the
build system and remove the old audioresample from the build system.
Fixes bug #558124, #385061, #346218, #116051.

gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_offset), (gst_base_audio_src_create):
Avoid nasty int overflows after about 12 hours and 25 minutes when these
code paths are triggered.
A free beer to Håvard Graff for finding this!

gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on

Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect):
A successful gst_poll_wait() doesn't always mean successful connect() on
Windows.  We should check errors by calling gst_poll_fd_has_error().
See #561924.

tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.

Original commit message from CVS:
* tests/check/elements/speexresample.c: (test_pipeline):
Make unit test again faster to prevent timeouts with valgrind.

Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
Fix typo in the docs.

Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
If no stream was found before receiving EOS, post an error message.
Fixes #561924.

Original commit message from CVS:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c: (gst_theora_enc_init),
(theora_buffer_from_packet), (theora_push_packet),
(theora_enc_sink_event), (theora_enc_is_discontinuous),
(theora_enc_chain):
Parse segment events.
Pass incomming buffer timestamps to outgoing buffers.
Use the running_time to construct the granulepos.
Fixes #562163.

Original commit message from CVS:
* gst/playback/gstplaybin2.c: (activate_group):
Fix buffer-duration property.

gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Really fix audiosink drain handling by keeping track of the running_time
of the last sample.

gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.

Original commit message from CVS:
* gst/playback/gstplaybin2.c:
Add notification of current stream. Add ability to configure buffer
sizes.
* gst/playback/gsturidecodebin.c:
Add ability to configure buffer sizes for streaming mode.
Bug #561734.

gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Time is already in running_time. Remove base_time handling. Fixes
audiosinks not draining and thus chopping some audio in the end.

Original commit message from CVS:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
If we're muxing a dirac stream, flush the page after every picture.

gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Add one log message to check for audio_drained. Sync one log message
with the condition. Send EOS after draining audio in pull mode.

ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr...

Original commit message from CVS:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
Use gst_buffer_try_new_and_alloc() and fail properly if the
allocation failed. This prevents abort() if downstream elements
request an insane amount of memory.