+2009-09-11 22:09:06 +0200 Benjamin Otte <otte@gnome.org>
+
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix crashes with even widths
+ The fix for green lines introduced by commit
+ 35fdfcc6258c66ba462a4330a35deffb0f2b501d caused invalid memory accesses
+ for even widths. This patch fixes it.
+
+2009-09-11 15:11:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ playbin2: Implement GstStreamVolume interface
+
+2009-09-11 15:04:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/volume/gstvolume.c:
+ * gst/volume/gstvolume.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/volume.c:
+ volume: Implement GstStreamVolume interface
+
+2009-09-11 14:54:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * docs/libs/gst-plugins-base-libs-docs.sgml:
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/interfaces/streamvolume.c:
+ * gst-libs/gst/interfaces/streamvolume.h:
+ * gst/playback/Makefile.am:
+ * win32/common/libgstinterfaces.def:
+ interfaces: API: Add GstStreamVolume interface
+ Fixes bug #567660.
+
+2009-09-11 12:20:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: properly fix the HTTP manual mode
+ When we're not parsing HTTP, return EPARSE when we get an HTTP
+ message.
+
+2009-09-11 10:16:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/interfaces/mixertrack.h:
+ mixertrack: add READONLY and WRITEONLY flags
+ Should really have been READABLE and WRITABLE, but those are hard to
+ add whilst maintaining backwards compatibility. See #343615.
+ API: GST_MIXER_TRACK_READONLY
+ API: GST_MIXER_TRACK_WRITEONLY
+
+2009-09-11 10:02:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: fix build against core that has debugging disabled
+ The macro is called GST_DISABLE_GST_DEBUG, not GST_DISABLE_DEBUG.
+
+2009-09-11 07:38:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videorate/gstvideorate.c:
+ videorate: Add Since marker for the new skip-to-first property
+
+2009-09-11 07:36:10 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/videorate/gstvideorate.c:
+ * gst/videorate/gstvideorate.h:
+ videorate: Make videorate work with a live source
+ Add a property that makes videorate skip to the first buffer it
+ receives instead of padding the stream from segment start to the
+ first real buffer.
+ Fixes bug #567928.
+
+2009-09-11 07:20:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/fft/gstfft.h:
+ * gst-libs/gst/fft/gstfftf32.h:
+ * gst-libs/gst/fft/gstfftf64.h:
+ * gst-libs/gst/fft/gstffts16.h:
+ * gst-libs/gst/fft/gstffts32.h:
+ fft: Mark one function as const and add notes that the structs should be private in 0.11
+
+2009-09-10 22:28:19 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: add human readable format names when logging
+ Add string array with human readable names for format and type to be used in log
+ statements.
+
+2009-09-10 18:19:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppay: don't print RTP timestamps as clocktime
+ Don't try to print the RTP timestamp as a GstClockTime, it's just a guint32.
+ Fixes #594757
+
+2009-09-10 16:55:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstplaybin.c:
+ * gst/playback/gstplaybin2.c:
+ playbin(2): Document that the volume property uses a linear scale
+ Fixes bug #571610.
+
+2009-09-10 14:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: don't return EPARSE
+ Don't blindly return EPARSE when http mode is disabled.
+ Restore old http mode after temporarily setting it to TRUE.
+
+2009-09-10 12:38:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: add ugly backward compat hack
+ Check for pulsesink < 0.10.17 because it includes code that is now included in
+ baseaudiosink. Disable that code in baseaudiosink to be compatible with the
+ older version.
+
+2009-09-10 10:56:29 +0200 Benjamin Otte <otte@gnome.org>
+
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Handle YVU9/YUV9 conversion with odd widths
+ A green border could be visible when converting to Y444 or RGB, because
+ the last chroma samples weren't copied correctly
+
+2009-09-10 10:43:37 +0200 Benjamin Otte <otte@gnome.org>
+
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix YVU9 and YUV9
+ - Buffer sizes were computed different from ffmpegcolorspace
+ - Green bar on right size for widths not divisable by 4
+
+2009-09-10 10:08:28 +0200 Benjamin Otte <otte@gnome.org>
+
+ * gst/videotestsrc/videotestsrc.c:
+ videotestsrc: Fix image for odd widths in some formats
+ videotestsrc rounds chroma down. This causes it to omit the last chroma
+ value completely for odd widths when the chroma is downsampled.
+ This patch special cases the last pixel to not be rounded down.
+
+2009-09-10 10:02:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ oggdemux: Handle kate and cmml as sparse streams too
+
+2009-09-10 10:00:16 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggdemux.h:
+ oggdemux: Better handling of sparse streams by sending segment updates
+ Fixes bug #397419.
+
+2009-09-10 09:43:28 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/playback/gsturidecodebin.c:
+ docs: tell a biit more about uri-decodebin and buffering
+
+2009-09-09 18:24:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: take clock time in setcaps
+ Take the time of the clock so that the last_time field is set. This is important
+ for sinks that restart their internal ringbuffer after a caps change and need to
+ know the last know position.
+
+2009-09-09 18:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudioclock.c:
+ audioclock: add some more debug
+
+2009-09-09 16:44:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/theora/theoraenc.c:
+ theoraenc: Print a debug message with supported formats
+
+2009-09-07 17:29:38 +0200 Benjamin Otte <otte@gnome.org>
+
+ * ext/theora/theoraenc.c:
+ theora: Check supported input formats in getcaps function
+ We want to fail early when an older libtheora release is used that does
+ not support Y444 or Y42B formats, so use a getcaps function that does
+ this.
+
+2009-09-04 21:37:04 +0200 Benjamin Otte <otte@gnome.org>
+
+ * ext/theora/theoraenc.c:
+ theora: Implement support in theoraenc for Y444 and Y42B
+ Fixes bug #594165.
+
+2009-09-04 20:23:52 +0200 Benjamin Otte <otte@gnome.org>
+
+ * ext/theora/theoraenc.c:
+ theora: Refactor the buffer copy code
+
+2009-09-04 16:59:49 +0200 Benjamin Otte <otte@gnome.org>
+
+ * ext/theora/theoraenc.c:
+ theora: Split yuv_buffer creation into its own function
+
+2009-09-04 16:49:08 +0200 Benjamin Otte <otte@gnome.org>
+
+ * ext/theora/theoraenc.c:
+ theora: Split out buffer resize in its own function
+
+2009-09-04 14:06:09 +0200 Benjamin Otte <otte@gnome.org>
+
+ * ext/theora/theoraenc.c:
+ theora: Add assertions that functions don't fail
+ Some functions in libtheora can return an error, but that error cannot
+ ever happen inside theoraenc. In those cases assert that it doesn't.
+
+2009-09-09 16:21:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/seek/seek.c:
+ seek: make stop state configurable
+ Make it easy to experiment with different stop states (NULL and READY)
+
+2009-09-09 16:19:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosink.c:
+ baseaudiosink: correct for clock reset
+ When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
+ also make sure that the clock is updated with the elapsed time so that it
+ alsways increments even when the ringbuffer goes back to 0. When this happened
+ we need to adjust the sample position for the reset ringbuffer.
+ Fixes #594136
+
+2009-09-09 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosink.h:
+ baseaudiosink: whitespace fixes
+
+2009-09-09 16:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: add more debug
+
+2009-09-09 10:25:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/interfaces/colorbalance.h:
+ * gst-libs/gst/interfaces/mixer.h:
+ whitespace fixes
+
+2009-09-08 17:59:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/video/gstvideosink.c:
+ * gst-libs/gst/video/gstvideosink.h:
+ videosink: add "show-preroll-frame" property
+ Add a property to disable rendering of video frames during preroll. This
+ will only work for videosinks that use the new ::show_frame() vfunc instead
+ of overriding basesink's preroll and render vfuncs directly.
+ API: GstVideoSink:show-preroll-frame
+
+2009-09-08 17:43:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * sys/ximage/ximagesink.c:
+ * sys/xvimage/xvimagesink.c:
+ ximagesink, xvimagesink: use new GstVideoSink::show_frame() vfunc
+
+2009-09-08 18:19:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/video/gstvideosink.c:
+ * gst-libs/gst/video/gstvideosink.h:
+ video: add GstVideoSinkClass::show_frame()
+ Add ::show_frame() vfunc which maps to basesink's ::preroll and ::render
+ vfuncs and add some gtk-doc chunks.
+ API: GstVideoSinkClass::show_frame()
+
+2009-09-08 16:00:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: don't do stuff inside g_return_val_if_fail() statements
+ Or it will all fall apart if someone compiles with -DG_DISABLE_ASSERT.
+
+2009-08-31 20:24:22 +0200 Havard Graff <havard.graff@tandberg.com>
+
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: Fix compiler warning with MSVC
+ Fixes bug #594275.
+
+2009-08-31 20:31:56 +0200 Havard Graff <havard.graff@tandberg.com>
+
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ basertpdepayload: fix event forwarding
+
+2009-08-31 20:36:37 +0200 Havard Graff <havard.graff@tandberg.com>
+
+ * gst-libs/gst/rtp/gstrtcpbuffer.c:
+ rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
+ Fixes #594258
+
+2009-09-08 13:02:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gstplaysink.h:
+ fix whitespace
+
+2009-09-08 12:59:20 +0200 Håvard Graff <havard.graff@tandberg.com>
+
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: improve slave skew resync
+ The old one did the mistake of not actually advancing the ringbuffer, it just
+ adjusted the segbase, introducing the whole lenght of the ringbuffer as an
+ extra delay in the pipeline.
+ Also make sure that the resync can never go back in time, producing the same
+ timestamps that has already been produced, as this can cause severe problems
+ for sinks and other synching mechanisms.
+ Fixes #594256
+
+2009-09-07 17:13:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: disable typefinder for headerless flac
+ Disable headerless flac typefinder as long as it happily typefinds anything
+ including /dev/urandom as flac and as long as it's not particularly useful
+ given that such streams don't really exist in the wild.
+ Also fix up some comments so that gtk-doc doesn't complain about them.
+
+2009-09-06 15:21:43 +0300 René Stadler <mail@renestadler.de>
+
+ * sys/ximage/ximagesink.c:
+ ximagesink: fix small memory leak when setting window title
+
+2009-09-06 01:42:42 +0300 René Stadler <mail@renestadler.de>
+
+ * sys/xvimage/xvimagesink.c:
+ xvimagesink: fix small memory leak when setting window title
+
+2009-09-05 13:55:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * .gitignore:
+ introspection: Add *.gir and *.typelib to .gitignore
+
+2009-09-05 13:46:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ introduction: Fix out-of-tree build
+
+2009-09-05 13:13:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/Makefile.am:
+ rtsp: Fix introspection build by ordering sources/headers in dependency order
+
+2009-09-05 13:09:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Remove debug echo
+
+2009-09-05 13:08:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Fix build of introspection data by using dependency order for the headers/sources
+
+2009-09-05 12:31:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/cdda/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ introspection: Strip Gst prefix from all types/functions
+
+2009-09-05 11:49:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/Makefile.am:
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/fft/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/netbuffer/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ * gst-libs/gst/riff/Makefile.am:
+ * gst-libs/gst/rtp/Makefile.am:
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/sdp/Makefile.am:
+ * gst-libs/gst/tag/Makefile.am:
+ * gst-libs/gst/video/Makefile.am:
+ introspection: Fix build if gir-repository is not installed
+
+2009-09-05 11:37:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/video/Makefile.am:
+ video: Add gobject-introspection support
+
+2009-09-05 11:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/tag/Makefile.am:
+ tag: Add gobject-introspection support
+
+2009-09-05 11:34:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/sdp/Makefile.am:
+ sdp: Add gobject-introspection support
+
+2009-09-05 11:31:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/app/Makefile.am:
+ * gst-libs/gst/audio/Makefile.am:
+ * gst-libs/gst/interfaces/Makefile.am:
+ * gst-libs/gst/pbutils/Makefile.am:
+ libs: Add nodist headers and sources to the introspection files
+
+2009-09-05 11:28:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/Makefile.am:
+ rtsp: Add gobject-introspection support
+
+2009-09-05 11:25:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/rtp/Makefile.am:
+ rtp: Add gobject-introspection support
+
+2009-09-05 11:23:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/riff/Makefile.am:
+ riff: Add gobject-introspection support
+
+2009-09-05 11:20:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/pbutils/Makefile.am:
+ pbutils: Add gobject-introspection support
+
+2009-09-05 11:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/netbuffer/Makefile.am:
+ netbuffer: Add gobject-introspection support
+
+2009-09-05 11:15:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/interfaces/Makefile.am:
+ interfaces: Add gobject-introspection support
+
+2009-09-05 11:04:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/fft/Makefile.am:
+ fft: Add gobject-introspection support
+
+2009-09-05 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/cdda/Makefile.am:
+ cdda: Add gobject-introspection support
+ This is disabled for now until gobject-introspection is fixed
+
+2009-09-05 10:50:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/audio/Makefile.am:
+ audio: Add gobject-introspection support
+
+2009-09-05 10:40:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ * gst-libs/gst/app/Makefile.am:
+ app: Add gobject-introspection support
+
+2009-09-05 10:20:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 00a859e to 19fa4f3
+
+2009-09-04 15:48:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: fix midi typefinding
+ We already have a audio/midi typefinder so don't override it with the midi in
+ RIFF typefinder or else we fail to detect plain midi files.
+
+2009-09-04 11:29:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: do buffering for more uris
+ Add ssh://, ftp://, sftp://, myth:// to the list of uris that require
+ buffering.
+ Fixes #594020
+
+2009-09-04 07:36:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add typefinder for Midi inside RIFF
+ This is a standard Midi file format that should be supported by
+ all Midi decoders and also has the mimetype audio/mid according to
+ the Midi specification homepage.
+ Fixes bug #594094.
+
+2009-09-03 18:53:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: add some debugging
+
+2009-09-03 17:53:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: handle gaps
+ Add various conversion functions between time<->bytes<->rtptime that will be
+ used later on.
+ Refactor the min/max packet length code so that it can be used for both
+ sample/frame based payloaders. Cache the returned values.
+ code cleanups.
+ When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
+ same gap as the GStreamer timestamps gap.
+
+2009-09-03 14:13:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: fix frame duration calculations
+ Fix the calculation of the frame duration and rtp timestamps.
+ Add some debugging
+
+2009-09-03 14:13:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ rtppay: add some debugging
+
+2009-09-02 19:49:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: use offsets for RTP timestamps
+ Have a custom sample/frame function to generate an offset that the base class
+ will use for generating RTP timestamps. This results in perfect RTP timestamps
+ on the output buffers.
+ Refactor setting metadata on output buffers.
+ Add some more functionality to _flush().
+ Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
+ the next outgoing buffer.
+ Flush the pending data on EOS.
+
+2009-09-02 13:13:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: move function around
+
+2009-09-02 13:12:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: fix sample duration calculation
+
+2009-09-02 12:24:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppay: more refactoring
+ Unify the sample/frame buffer handling code by making the functions plugable.
+
+2009-09-02 12:03:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ audiortppayload: refactor some more
+ Refactor getting the packet min/max size and alignment code.
+ Refactor converting bytes to time.
+ change some variable to something shorter.
+
+2009-09-02 10:46:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
+ * win32/common/libgstrtp.def:
+ audiortppayload: refactor and cleanup
+ Always use the adapter when we need to fragment the incomming buffer. Use more
+ modern adapter functions to avoid malloc and memcpy. The overall result is that
+ the code looks cleaner while it should be equally fast and in some case avoid a
+ memcpy and malloc.
+ Use the adapter timestamping functions for more precise timestamps in case of
+ weird disconts.
+ Cache some values instead of recalculating them.
+ Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
+ the internal adapter.
+ API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
+
+2009-09-03 16:56:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Update common
+
+2009-09-03 11:29:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppay: add property to disable perfect RTP time
+ Add a property to disable the generation of perfect RTP timestamps. By default
+ it is active.
+ API: GstBaseRTPPayload::perfect-rtptime
+
+2009-09-02 19:47:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ basertppay: allow subclasses to influence RTP time
+ Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
+ which RTP timestamps are generated. Usually timestamps are created from the
+ GStreamer timestamps on the buffer, which could result in imperfect RTP
+ timestamps.
+
+2009-09-02 19:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppay: add macro to cast
+
+2009-09-01 18:26:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiopayload: code cleanups
+
+2009-09-01 18:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+ audiortppayload: don't check adapter
+ the adapter is never NULL so we don't need to check it.
+ Use _scale functions to avoid overflows.
+
+2009-09-03 00:14:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * gst/typefind/Makefile.am:
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: move gio-based xdg mime typefinder from -bad to -base
+ Its purposes is mainly to avoid false positives (e.g. mp3 typefinder
+ reporting a 20% probability and somesuch). Won't be registered if
+ the gio plugin has been disabled via ./configure --disable-gio.
+
+2009-09-01 15:06:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: GstAdapter is not a GstObject and should be freed with g_object_unref
+
+2009-09-01 15:02:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * sys/v4l/v4lsrc_calls.c:
+ v4lsrc: fix timestamping for when we do not have a clock yet
+ Should fix #559049.
+
+2009-09-01 14:30:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * sys/v4l/v4lsrc_calls.c:
+ v4lsrc: don't log not-yet-initialised integer value
+
+2009-09-01 14:28:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * sys/v4l/v4lsrc_calls.c:
+ v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize
+ And reflow code to be more indent friendly.
+
+2009-09-01 10:39:52 +0200 Jonas Holmberg <jonas.holmberg@axis.com>
+
+ * gst-libs/gst/rtp/gstbasertppayload.c:
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppayload: Make instance init faster by not reading /dev/urandom 3 times
+ ... which is the default seed when creating a new GRand. Because
+ GLib in older versions used buffered IO this would take a lot of time.
+ Instead use the global GRand for getting random numbers and keep the
+ three instance GRand for backward compatibility with a simple seed.
+ Fixes bug #593284.
+
+2009-08-31 22:48:01 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/adder/gstadder.c:
+ adder: improve caps filter functionality. Fixes #590146.
+ Also use the capsfilter if there is no src-peer as the caps constrain what
+ we can do. Don't create any_caps as a default, as we check for NULL to skip the
+ filtering. This is a (small) performance regression as we always intersect
+ otherwise.
+
+2009-08-31 11:10:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: Post missing plugin messages before any error messages
+
+2009-08-28 19:06:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ cddabasesrc: safely handle the indexes
+
+2009-08-28 19:06:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * win32/common/libgstrtsp.def:
+ def: add new rtsp symbols
+
+2009-08-28 14:08:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtp/gstbasertppayload.h:
+ basertppayload: whitespace fixes.
+
+2009-08-27 18:59:49 +0200 Marc-André Lureau <mlureau@flumotion.com>
+
+ * gst/gdp/gstgdppay.c:
+ Bug 593035 - set IN_CAPS for streamheader buffer
+
+2009-08-26 16:56:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ playbin: The internally linked pad of the selector might be NULL in some cases
+
+2009-08-26 16:45:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ playbin: Fix iterate internal linked pads functions for the stream selectors
+ This now used the new gst_iterator_new_single() function and as a side effect
+ fixes bug #592864.
+
+2009-08-26 09:08:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/riff/riff-ids.h:
+ * gst-libs/gst/riff/riff-read.c:
+ riff: Add support for AVF files
+ AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF.
+ Fixes bug #593117.
+
+2009-08-26 09:08:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Detect AVF files as RIFF files too
+ AVF is valid RIFF but has AVF0 as first fourcc instead of RIFF.
+ Partially fixes bug #593117.
+
+2009-08-21 11:51:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * tests/check/elements/audioresample.c:
+ audioresample: Add unit test for checking for timestamp drifts
+ This also checks for perfect timestamping and offsetting.
+
+2009-08-21 10:11:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Fix drain processing
+ In case we have to convert internally don't process output length input samples
+ but history length input samples.
+
+2009-08-21 10:02:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * tests/check/elements/audioresample.c:
+ audioresample: Improve debugging a bit in the unit test
+
+2009-08-21 10:00:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: On the first buffer we need discont handling
+ Otherwise we won't get upstream timestamps and everything and all
+ output buffers would have -1 timestamps.
+
+2009-08-21 08:23:39 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
+
+ * configure.ac:
+ * gst/subparse/gstsubparse.c:
+ subparse: Remove dependency on regex.h as it's not used anyway
+ Fixes bug #592544.
+
+2009-08-21 06:58:31 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
+
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Fix buffer overflow when pushing the drain
+
+2009-08-21 06:57:58 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
+
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ audioresample: Fix timestamp drift
+ Fixes bug #591934.
+
+2009-08-24 11:34:35 -0700 David Schleef <ds@schleef.org>
+
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/ogg/gstogmparse.c:
+ * ext/pango/gsttextrender.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ * gst/subparse/gstsubparse.c:
+ * sys/v4l/gstv4lmjpegsink.c:
+ * sys/v4l/gstv4lmjpegsrc.c:
+ * sys/v4l/gstv4lsrc.c:
+ Remove Ronald Bultje from Authors field
+ Replaced with "GStreamer maintainers
+ <gstreamer-devel@lists.sourceforge.net>" or just removed,
+ depending on the number of other authors.
+
+2009-08-24 15:06:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/playback/gstplaybin2.c:
+ playbin2: fix refcounting of _get_sink()
+ g_value_set_object() increases the refcount of the sink, which is not needed
+ because the object should already be refcounted. Make sure this is always the
+ case and use g_value_take_object().
+ Fixes: #592884
+
+2009-08-24 14:39:16 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ rtsp: Mark Transport as supporting multiple values.
+
+2009-08-24 13:58:17 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ rtsp: Added missing Since tags.
+
+2009-08-24 13:27:55 +0200 Eero Nurkkala <ext-eero.nurkkala at nokia.com>
+
+ * gst-libs/gst/audio/gstringbuffer.c:
+ ringbuffer: Improve audiosink startup performance
+ When we start the ringbuffer, immediatly continue processing samples if the
+ writer prepared some for us.
+ Fixes #545807
+
+2009-08-17 11:53:43 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Added new API for sending using GstRTSPWatch.
+ The new API to send messages using GstRTSPWatch will first try to send the
+ message immediately. Then, if that failed (or the message was not sent
+ fully), it will queue the remaining message for later delivery. This avoids
+ unnecessary context switches, and makes it possible to keep track of
+ whether the connection is blocked (the unblocking of the connection is
+ indicated by the reception of the message_sent signal).
+ This also deprecates the old API (gst_rtsp_watch_queue_data() and
+ gst_rtsp_watch_queue_message().)
+ API: gst_rtsp_watch_write_data()
+ API: gst_rtsp_watch_send_message()
+
+2009-08-17 11:46:32 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Made gst_rtsp_watch_queue_data() thread safe.
+
+2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Added gst_rtsp_connection_set_http_mode().
+ With gst_rtsp_connection_set_http_mode() it is possible to tell the
+ connection whether to allow HTTP messages to be supported. By enabling HTTP
+ support the automatic HTTP tunnel support will also be disabled.
+ API: gst_rtsp_connection_set_http_mode()
+
+2009-06-16 19:35:23 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Allow gst_rtsp_connection_do_tunnel() to just setup decoding context.
+ If the second connection passed to gst_rtsp_connection_do_tunnel() is NULL
+ then just setup the base64 decoding context for the first connection.
+
+2009-06-16 19:04:54 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Write as much as possible in gst_rtsp_source_dispatch().
+ Try to write as much as possible if there are multiple messages queued.
+
+2009-06-16 18:38:02 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Add error_full callback to GstRTSPWatchFuncs.
+ The error_full callback is similar to the error callback, but allows for
+ better error handling. For read errors a partial message is provided to
+ help an RTSP server generate a more correct error response, and for write
+ errors the write queue id of the failed message is returned.
+
+2009-08-17 18:29:17 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Made read_line() support LWS.
+ Rewrote read_line() to support LWS (Line White Space), the method used by
+ RTSP (and HTTP) to break long lines. Also added support for \r and \n as
+ line endings (in addition to the official \r\n).
+
+2009-08-20 14:12:50 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: Do not split headers which should not be split.
+ From RFC 2068 section 4.2: "Multiple message-header fields with the same
+ field-name may be present in a message if and only if the entire
+ field-value for that header field is defined as a comma-separated list
+ [i.e., #(values)]." This means that we should not split other headers which
+ may contain a comma, e.g., Range and Date.
+
+2009-08-20 14:12:09 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Parse WWW-Authenticate headers correctly.
+ Due to the odd syntax for WWW-Authenticate (and Proxy-Authenticate) which
+ allows commas both to separate between multiple challenges, and within the
+ challenges themself, we need to take some extra care to split these headers
+ correctly.
+
+2009-06-17 21:46:27 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Improve parse_line().
+ Make parse_line() handle keys with multiple values on one line correctly.
+
+2009-06-17 23:15:23 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Rewrote setup_tunneling().
+ Rewrote setup_tunneling() to use normal GstRTSPMessages instead of hard
+ coded strings and duplicates of the message parsing code.
+
+2009-08-24 10:20:16 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: Rewrote gen_tunnel_reply().
+ Rewrote gen_tunnel_reply() to generate a normal GstRTSPMessage rather
+ than a hard coded string.
+
+2009-08-24 10:19:35 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Ignore the Content-Length for POST requests.
+ The Content-Length for POST requests with an x-sessioncookie header should
+ be ignored as the length is bogus and only there to fool proxies.
+
+2009-06-17 20:52:48 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Normalize lines (remove extra whitespace) before parsing.
+
+2009-06-10 13:11:31 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Made parse_string() return a result.
+ This will catch parsing errors when a too long string is received.
+
+2009-06-10 11:43:31 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Improved parsing of messages.
+ Do not abort message parsing as soon as there is an error. Instead parse
+ as much as possible to allow a server to return as meaningful an error as
+ possible.
+
+2009-06-09 17:54:20 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ rtsp: Added support for HTTP messages
+
+2009-06-09 16:22:17 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp: Added gst_rtsp_connection_create_from_fd().
+ API: gst_rtsp_connection_create_from_fd()
+
+2009-06-09 15:27:17 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Add initial buffer support.
+ The initial buffer contains data for a connection which should be used
+ before starting to actually read anything from the socket.
+
+2009-08-24 13:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/app/gstappsink.c:
+ appsink: don't block in paused
+ When we are asked to unlock we should either leave the render function or call
+ the wait_preroll method to release the stream lock.
+ Fixes #592657
+
+2009-08-24 13:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ docs: fix includes for appsrc/appsink
+
+2009-08-24 11:24:27 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspdefs.c:
+ * gst-libs/gst/rtsp/gstrtspdefs.h:
+ rtsp: Add support for the Authentication-Info header.
+ The Authentication-Info header is defined in RFC 2617 (Digest Access
+ Authentication).
+
+2009-08-20 13:11:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/ogg/gstoggmux.c:
+ * tests/check/pipelines/oggmux.c:
+ oggmux: don't drop the streamheader field from the output caps
+ Revert previous 'fix' for bug #588717 and fix it properly, whilst
+ maintaining the streamheader field on the output caps. Also make
+ sure we don't leak header buffers we couldn't push when downstream
+ is unlinked. Add unit test for the presence of the streamheader
+ field on the output caps and for the issue from bug #588717.
+
+2009-08-18 21:45:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ streamselector/inputselector: Use iterate internal links instead of deprecated get internal links
+
+2009-08-19 09:31:51 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Avoid duplicated headers.
+ Remove any existing Session and Date headers before adding new ones
+ when sending a request. This may happen if the user of this code reuses
+ a request (rtspsrc does this when resending after authorization fails).
+
+2009-08-18 16:49:58 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: Corrected the HTTP digest authorization computation.
+ Do not use sizeof() on an array passed as an argument to a function and
+ expect to get anything but the size of a pointer. As a result only the
+ first 4 (or 8) bytes of the response buffer were initialized to 0 in
+ auth_digest_compute_response() which caused it to return a string which
+ was not NUL-terminated...
+
+2009-08-18 11:15:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Also send SEEK events directly to a subpicture sink
+
+2009-08-18 08:39:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ playsink: If a custom text sink is used, send events to it too
+ Before, SEEK events would be sent to the video sink, which wouldn't
+ be linked in any way to the subtitle part of the pipeline and
+ subparse would never see the SEEK event. This would then seek
+ the audio/video but the subtitles would continue from the old
+ position instead.
+ Fixes bug #591664.
+
+2009-08-18 08:20:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Make missing plugins emit a warning message, not an error message
+ The problem with an error message is, that it will stop playback completely
+ while it could be that only a audio decoder plugin is missing and the video
+ could be played with the available plugins.
+ See bug #591677.
+
+2009-08-13 17:42:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Post a correct error message for unknown types
+ Before we had STREAM/WRONG_TYPE but it's really CORE/MISSING_PLUGIN
+ because a plugin is missing and nothing else is wrong.
+ Also make it an error instead of a warning.
+ Really fixes bug #591677.
+
+2009-08-13 15:48:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gsturidecodebin.c:
+ uridecodebin: Post a missing plugin message additional to the error message on unknown types
+ Fixes bug #591677.
+
+2009-08-13 10:59:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/playback/gstplaysink.c:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/en_GB.po:
+ * po/es.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ playbin2: fix error message string
+ Fixes #591577.
+
+2009-08-05 15:38:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst-libs/gst/riff/riff-read.c:
+ riff: align API doc of gst_riff_parse_chunk with reality
+
+2009-08-05 15:36:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin2: avoid assertion failure on empty/NULL caps
+
+2009-08-12 12:09:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Also detect SVG by the <svg> starting tag
+ Not all SVG images have the DOCTYPE specified.
+
+2009-08-10 20:18:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: don't use GLib-2.18 function
+ g_checksum_reset() was added only in GLib 2.18, but we still require
+ only 2.16, so work around that if we only have 2.16. Fixes #591357.
+
+2009-08-10 15:40:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * tests/check/pipelines/streamheader.c:
+ streamheader: Fix caps leak in the vorbisenc unit test
+
+2009-08-10 14:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/check/pipelines/streamheader.c:
+ checks: fix stream header unit test hanging in gst_task_cleanup_all()
+ Set pipelines to NULL state and unref when done.
+
+2009-08-10 10:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/Makefile.am:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/md5.c:
+ * gst-libs/gst/rtsp/md5.h:
+ rtsp: Use GLib's GChecksum instead of our own MD5 implementation
+
+2009-08-10 03:46:39 +0300 Mart Raudsepp <leio@gentoo.org>
+
+ * gst-libs/gst/interfaces/navigation.c:
+ navigation: Fix doc blurb typo for gst_navigation_send_key_event
+
+2009-08-09 12:13:16 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: Allow . instead of , as millisecond delimiter in srt subtitles
+ Fixes bug #591207.
+
+2009-08-08 17:51:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gststreamselector.c:
+ Revert inlines that cause compiler warnings and are not needed anyway
+
+2009-08-08 15:54:57 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst-libs/gst/audio/gstaudioclock.c:
+ * gst-libs/gst/audio/gstaudiosink.c:
+ * gst-libs/gst/audio/gstaudiosrc.c:
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ * gst-libs/gst/audio/gstringbuffer.c:
+ * gst-libs/gst/interfaces/propertyprobe.c:
+ * gst-libs/gst/riff/riff-media.c:
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ * gst-libs/gst/video/gstvideofilter.c:
+ * gst-libs/gst/video/gstvideosink.c:
+ gst-libs: Remove dead assignments and resulting unused variables.
+
+2009-08-08 15:54:41 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * ext/alsa/gstalsadeviceprobe.c:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/gnomevfs/gstgnomevfssrc.c:
+ * ext/ogg/gstoggaviparse.c:
+ * ext/ogg/gstoggdemux.c:
+ * ext/ogg/gstoggmux.c:
+ * ext/pango/gsttextrender.c:
+ * ext/vorbis/vorbisenc.c:
+ ext: Remove dead assignments and resulting unused variables.
+
+2009-08-08 15:54:02 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/adder/gstadder.c:
+ * gst/audioconvert/gstaudioconvert.c:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/playback/gstdecodebin.c:
+ * gst/playback/gstdecodebin2.c:
+ * gst/playback/gstfactorylists.c:
+ * gst/playback/gstinputselector.c:
+ * gst/playback/gstplaysink.c:
+ * gst/playback/gststreamselector.c:
+ * gst/tcp/gsttcpclientsink.c:
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/vs_image.c:
+ * gst/videotestsrc/gstvideotestsrc.c:
+ gst: Remove dead assignments and resulting unused variables
+
+2009-08-07 13:05:42 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * docs/design/draft-va.txt:
+ docs: add draft for generic introduction of video acceleration APIs idea
+
+2009-08-07 08:53:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ Revert "theora: Convert theoradec to libtheora 1.0 API"
+ This reverts commit f1e142ac9dcfb754d85357b9077d5aee48559dd9.
+ Temporarily revert until we have a workaround for debian/ubuntu
+ packaging failure (see http://bugs.debian.org/528710).
+
+2009-08-07 09:32:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add typefinders for many game sound console formats supported by gme
+ These are AY, GBS, GYM, KSS, SAP and VGM. SPC and NSF already had typefinders.
+
+2009-07-16 11:29:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * ext/ogg/gstoggmux.c:
+ oggmux: fix warning when we're not linked downstream and error out properly
+ Fix caps warning when there's no element linked downstream, and pass
+ not-linked flow return value correctly up the chain, so we error out
+ correctly. Fixes #588717.
+
+2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
+
+ * ext/theora/gsttheoradec.h:
+ * ext/theora/theoradec.c:
+ theora: Convert theoradec to libtheora 1.0 API
+
+2009-08-06 20:47:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/pango/gsttextrender.c:
+ textrender: Fix blitting of text over the output buffer and cairo painting
+
+2009-08-06 09:13:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/pango/gsttextrender.c:
+ textrender: Fix endianness problems (i.e. make it work again on big endian architectures)
+
+2009-07-31 14:27:28 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * tests/icles/test-colorkey.c:
+ colorkey-test: fix xsync error
+
+2009-07-06 23:06:50 +0300 Siarhei Siamashka <siarhei.siamashka@nokia.com>
+
+ * gst/ffmpegcolorspace/imgconvert.c:
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats
+
+2009-07-14 12:33:29 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/playback/gstplaysink.c:
+ playbin2: smarter sink selection. Fixes #588523
+ Don't do fallbacks if application specified a sink element. When doing the
+ fallback use configured default elements instead of hardcoded linux only
+ elements. Improve error messages accordingly.
+
+2009-08-06 12:18:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/playback/gstqueue2.c:
+ queue2: post error message when pausing task if so appropriate
+ If a downstream element returns an error while upstream has already
+ put all data into queue2 (including EOS), upstream will no longer
+ chain into queue2, so it is up to queue2 to perform some
+ EOS handling / message posting in such cases. See #589991.
+
+2009-08-06 12:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/audio/gstbaseaudiosrc.c:
+ baseaudiosrc: change default slave method
+ Set the default slave method to the much better skew slaving algortihm.
+
+2009-08-06 12:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pango/gsttextoverlay.c:
+ textoverlay: make buffer writable
+ Make the input buffer writable before changing its contents.
+
+2009-08-06 09:55:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefinding: fix postscript typefinder probability
+ Two bytes for a rare format hardly warrants MAXIMUM typefinding
+ probability, POSSIBLE seems more appropriate.
+
+2009-08-04 14:55:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/pango/gsttextoverlay.c:
+ pango: Send queries from the srcpad directly to the video sinkpad
+
+2009-08-04 14:32:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/subparse/gstsubparse.c:
+ subparse: Implement POSITION query
+
+2009-08-04 14:29:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/subparse/gstsubparse.c:
+ * gst/subparse/samiparse.c:
+ subparse: Implement SEEKING query
+
+2009-08-04 14:14:53 +0200 John Millikin <jmillikin@gmail.com>
+
+ * configure.ac:
+ * gst-libs/gst/tag/gstid3tag.c:
+ * gst-libs/gst/tag/gstvorbistag.c:
+ tag: Add support for ALBUM_ARTIST tag in vorbiscomments and ID3v2 tags
+ Require latest core for this.
+ Fixes bug #590430.
+
+2009-08-04 12:46:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ pango: Add support for xRGB and BGRx formats
+
+2009-08-04 12:22:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/pango/gsttextoverlay.c:
+ pango: Fix endianness issues from the pangocairo switch
+ cairo's ARGB is in native endianness, i.e. ARGB on big endian architectures
+ and BGRA on little endian architectures.
+
+2009-08-04 12:11:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/pango/gsttextoverlay.c:
+ pango: Re-add shading support which was dropped by a previous patch
+
+2009-08-04 11:58:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ * ext/pango/gsttextoverlay.c:
+ pango: Check if pangocairo supports vertical rendering and fix properties
+
+2009-08-04 11:45:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/pango/gsttextrender.c:
+ textrender: Use PROP_X instead of ARG_X consistently
+
+2009-08-04 11:42:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/pango/gstclockoverlay.c:
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttimeoverlay.c:
+ pango: Some minor cleanup
+
+2009-08-04 11:36:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ pango: Check for pangocairo instead of pangoft2
+
+2009-08-04 11:35:10 +0200 Young-Ho Cha <ganadist@chollian.net>
+
+ * ext/pango/gsttextoverlay.c:
+ * ext/pango/gsttextoverlay.h:
+ * ext/pango/gsttextrender.c:
+ * ext/pango/gsttextrender.h:
+ pango: Use pango-cairo instead of pango-ft2
+ pango-cairo will always use the native font rendering backend
+ of the platform and provides better results.
+ Fixes bug #340887.
+
+2009-08-04 10:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add SVG typefinder
+
+2009-08-04 10:29:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add postscript typefinder
+
+2009-07-30 15:08:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Use static caps again for MPEG4 typefinding
+
+2009-07-30 15:05:28 +0200 Arnout Vandecappelle <arnout@mind.be>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Implement better & more flexible MPEG4 typefinding
+ This detects more MPEG4 streams as MPEG4.
+ Fixes bug #556537.
+
+2009-07-30 14:04:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst-libs/gst/cdda/gstcddabasesrc.c:
+ cddabasesrc: Allow to specify the device name in the URI
+ The allowed URI scheme is now:
+ cdda://(device#)?track
+ Also allow every combination of uppercase and lowercase
+ characters for the protocol part.
+ Fixes bug #321532.
+
+2009-07-30 12:37:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Restrict width/height to 2^15 - 1
+ Otherwise integer overflows will happen, resulting in segmentation faults.
+ Fixes bug #590243.
+
+2009-07-29 14:55:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/ffmpegcolorspace/imgconvert_template.h:
+ ffmpegcolorspace: Fix indention of template header
+
+2009-07-29 14:10:35 +0200 Philip Jägenstedt <philipj@opera.com>
+
+ * gst-libs/gst/app/gstappsrc.c:
+ appsrc: Clarify documentation about caps and linkage
+ Fixes bug #589095.
+
+2009-07-29 07:42:05 +0200 Benjamin Gaignard <benjamin@gaignard.net>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Fix typefinding of SDP files
+ Fixes bug #589574.
+
+2009-07-28 20:50:06 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
+
+ * gst/audioresample/gstaudioresample.c:
+ audioresample: Take the output offsets from the input if possible
+ Fixes bug #588915.
+
+2009-07-28 15:54:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Make sure to allocate enough memory for the temporary buffer
+ and fix scaling of odd-height interlaced video.
+
+2009-07-28 15:18:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videoscale/gstvideoscale.c:
+ videoscale: Fix interlaced scaling for I420
+ ...and some other minor mistakes in the previous change.
+
+2009-07-28 14:12:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/ffmpegcolorspace/avcodec.h:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
+ * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
+ * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
+ * gst/ffmpegcolorspace/imgconvert.c:
+ ffmpegcolorspace: Include interlacing information in the AVPicture
+ This later allows to handle interlaced AVPicture different than
+ progressive ones which is needed for horizontally subsampled YUV
+ formats, see bug #589242.
+
+2009-07-28 13:55:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videoscale/gstvideoscale.c:
+ * gst/videoscale/gstvideoscale.h:
+ videoscale: Add support for interlaced content
+ videoscale is not mixing content of two seperate fields anymore
+ and does scaling on every field separately.
+ Fixes bug #588761.
+
+2009-08-06 01:44:24 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * configure.ac:
+ back to development -> 0.10.24.1
+
+2009-08-05 02:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst-plugins-base.doap:
+ Add 0.10.24 release to the doap file
+
=== release 0.10.24 ===
-2009-08-05 Jan Schmidt <jan.schmidt@sun.com>
+2009-08-05 00:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 0.10.24, "Counting the days"
+ * 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/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.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:
+ Release 0.10.24
2009-08-05 00:38:40 +0100 Jan Schmidt <thaytan@noraisin.net>
--- /dev/null
+# translation of gst-plugins-base.master.po to Basque
+# Copyright (C) 2004, 2005, 2006, 2008, 2009 Free Software Foundation, Inc.
+# This file is distributed under the same license as the gst-plugins-base package.
+#
+# Iñaki Larrañaga Murgoitio <dooteo@euskalgnu.org>, 2009.
+# Mikel Olasagasti Uranga <hey_neken@mundurat.net>, 2009.
+msgid ""
+msgstr ""
+"Project-Id-Version: gst-plugins-base-0.10.22.2\n"
+"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n"
+"POT-Creation-Date: 2009-09-11 23:31+0100\n"
+"PO-Revision-Date: 2009-08-29 12:55+0200\n"
+"Last-Translator: Mikel Olasagasti Uranga <hey_neken@mundurat.net>\n"
+"Language-Team: Basque <translation-team-eu@lists.sourceforge.net>\n"
+"MIME-Version: 1.0\n"
+"Content-Type: text/plain; charset=UTF-8\n"
+"Content-Transfer-Encoding: 8bit\n"
+"X-Generator: KBabel 1.11.4\n"
+"Plural-Forms: nplurals=2; plural=(n != 1);\n"
+
+msgid "Master"
+msgstr "Maisua"
+
+msgid "Bass"
+msgstr "Baxua"
+
+msgid "Treble"
+msgstr "Altua"
+
+msgid "PCM"
+msgstr "PCM"
+
+msgid "Synth"
+msgstr "Sintetizadorea"
+
+msgid "Line-in"
+msgstr "Linea-sarrera"
+
+msgid "CD"
+msgstr "CDa"
+
+msgid "Microphone"
+msgstr "Mikrofonoa"
+
+msgid "PC Speaker"
+msgstr "Ordenagailuko bozgorailua"
+
+msgid "Playback"
+msgstr "Erreproduzitzea"
+
+msgid "Capture"
+msgstr "Kapturatzea"
+
+msgid "Could not open device for playback in mono mode."
+msgstr "Ezin izan da gailua ireki mono moduan erreproduzitzeko."
+
+msgid "Could not open device for playback in stereo mode."
+msgstr "Ezin izan da gailua ireki estereo moduan erreproduzitzeko."
+
+#, c-format
+msgid "Could not open device for playback in %d-channel mode."
+msgstr "Ezin izan da gailua ireki %d kanaleko moduan erreproduzitzeko."
+
+msgid ""
+"Could not open audio device for playback. Device is being used by another "
+"application."
+msgstr ""
+"Ezin izan da audioaren gailua ireki erreproduzitzeko. Beste aplikazio batek "
+"darabil gailua."
+
+msgid "Could not open audio device for playback."
+msgstr "Ezin izan da audioaren gailua ireki erreproduzitzeko."
+
+msgid "Could not open device for recording in mono mode."
+msgstr "Ezin izan da gailua ireki mono moduan grabatzeko."
+
+msgid "Could not open device for recording in stereo mode."
+msgstr "Ezin izan da gailua ireki estereo moduan grabatzeko."
+
+#, c-format
+msgid "Could not open device for recording in %d-channel mode"
+msgstr "Ezin izan da gailua ireki %d kanaleko moduan grabatzeko."
+
+msgid ""
+"Could not open audio device for recording. Device is being used by another "
+"application."
+msgstr ""
+"Ezin izan da audioaren gailua ireki grabatzeko. Beste aplikazio batek "
+"darabil gailua."
+
+msgid "Could not open audio device for recording."
+msgstr "Ezin izan da audioaren gailua ireki grabatzeko."
+
+msgid "Could not open CD device for reading."
+msgstr "Ezin izan da CD gailua ireki irakurtzeko."
+
+msgid "Could not seek CD."
+msgstr "Ezin izan da CDan bilatu."
+
+msgid "Could not read CD."
+msgstr "Ezin izan da CDa irakurri."
+
+#, c-format
+msgid "Could not open vfs file \"%s\" for writing: %s."
+msgstr "Ezin izan da \"%s\" vfs fitxategia ireki idazteko: %s."
+
+msgid "No filename given"
+msgstr "Ez da fitxategi-izenik eman"
+
+#, c-format
+msgid "Could not close vfs file \"%s\"."
+msgstr "Ezin izan da itxi \"%s\" vfs fitxategia."
+
+#, c-format
+msgid "Error while writing to file \"%s\"."
+msgstr "Errorea gertatu da \"%s\" fitxategian idaztean."
+
+msgid "Internal data stream error."
+msgstr "Datu-korrontearen barne-errorea."
+
+#, c-format
+msgid "A %s plugin is required to play this stream, but not installed."
+msgstr ""
+"%s plugina behar da korronte hori erreproduzitzeko, baina ez dago instalatua."
+
+msgid "This appears to be a text file"
+msgstr "Testu-fitxategia dela dirudi"
+
+msgid "Could not determine type of stream"
+msgstr "Ezin izan da korronte mota zehaztu"
+
+#, c-format
+msgid "Invalid subtitle URI \"%s\", subtitles disabled."
+msgstr ""
+"Azpitituluen \"%s\" URIa ez da baliozkoa. Azpitituluak desgaitu egin dira."
+
+msgid "No URI specified to play from."
+msgstr "Ez da URIrik zehaztu irakurtzeko."
+
+#, c-format
+msgid "Invalid URI \"%s\"."
+msgstr "\"%s\" URI baliogabea."
+
+msgid "RTSP streams cannot be played yet."
+msgstr "Oraindik ezin dira erreproduzitu RTSP korronteak."
+
+msgid "Could not create \"decodebin\" element."
+msgstr "Ezin izan da sortu \"decodebin\" elementua."
+
+msgid "Source element is invalid."
+msgstr "Iturburuko elementua baliogabea da."
+
+msgid ""
+"Only a subtitle stream was detected. Either you are loading a subtitle file "
+"or some other type of text file, or the media file was not recognized."
+msgstr ""
+"Azpitituluen korrontea soilik detektatu da. Azpitituluen fitxategi bat edo "
+"beste testu-fitxategi bat ari zara kargatzen, edo multimedia-fitxategi hori "
+"ez da onartzen."
+
+msgid ""
+"You do not have a decoder installed to handle this file. You might need to "
+"install the necessary plugins."
+msgstr ""
+"Ez da instalatu fitxategi hori maneiatzeko deskodetzailerik Litekeena da "
+"beharrezko pluginak instalatu behar izatea."
+
+msgid "This is not a media file"
+msgstr "Hori ez da multimedia-fitxategi bat"
+
+msgid "A subtitle stream was detected, but no video stream."
+msgstr "Azpitituluen korrontea detektatu da, baina ez bideoarena."
+
+msgid "Both autovideosink and xvimagesink elements are missing."
+msgstr "Autovideosink eta xvimagesink elementuak, biak, falta dira."
+
+#, c-format
+msgid "Missing element '%s' - check your GStreamer installation."
+msgstr "'%s' elementua falta da. Begiratu GStreamer ondo instalatua dagoen."
+
+msgid "Both autoaudiosink and alsasink elements are missing."
+msgstr "Autoaudiosink eta alsasink elementuak, biak, falta dira."
+
+#, c-format
+msgid "Both autovideosink and %s elements are missing."
+msgstr "Autovideosink eta %s elementuak, biak, falta dira."
+
+msgid "The autovideosink element is missing."
+msgstr "Autovideosink elementua falta da."
+
+#, fuzzy, c-format
+msgid "Both autovideosink and %s elements are not working."
+msgstr "Autovideosink eta %s elementuak, biak, ez dira lanean ari."
+
+msgid "The autovideosink element is not working."
+msgstr "Autovideosink elementua ez da lanean ari."
+
+msgid "Custom text sink element is not usable."
+msgstr "Helburuko testu pertsonalizatua ez dago erabilgarri."
+
+msgid "No volume control found"
+msgstr "Ez da bolumenaren kontrolik aurkitu"
+
+#, c-format
+msgid "Both autoaudiosink and %s elements are missing."
+msgstr "Autoaudiosink eta %s elementuak, biak, falta dira."
+
+msgid "The autoaudiosink element is missing."
+msgstr "Autoaudiosink falta da."
+
+#, fuzzy, c-format
+msgid "Both autoaudiosink and %s elements are not working."
+msgstr "Autoaudiosink eta %s elementuak, biak, ez dira lanean ari."
+
+msgid "The autoaudiosink element is not working."
+msgstr "Autoaudiosink elementua ez da lanean ari."
+
+msgid "Can't play a text file without video."
+msgstr "Ezin da bideorik gabeko testu-fitxategi bat erreproduzitu."
+
+#, fuzzy
+msgid "Can't display both text subtitles and subpictures."
+msgstr "Ezin dira testuaren azpititulu eta azpiirudiak bistaratu."
+
+msgid "No Temp directory specified."
+msgstr "Ez da aldi baterako direktoriorik zehaztu."
+
+#, c-format
+msgid "Could not create temp file \"%s\"."
+msgstr "Ezin izan da aldi baterako \"%s\" fitxategia sortu."
+
+#, c-format
+msgid "Could not open file \"%s\" for reading."
+msgstr "Ezin izan da \"%s\" fitxategia ireki irakurtzeko."
+
+msgid "Internal data flow error."
+msgstr "Datu-korrontearen barne-errorea."
+
+#, c-format
+msgid "No decoder available for type '%s'."
+msgstr "Ez dago deskodetzaile erabilgarririk '%s' motarentzat."
+
+msgid "This stream type cannot be played yet."
+msgstr "Oraindik ezin da erreproduzitu korronte mota hori."
+
+#, c-format
+msgid "No URI handler implemented for \"%s\"."
+msgstr "Ez da URI maneiatzailerik ezarri \"%s\"(r)entzat."
+
+msgid "Could not create \"decodebin2\" element."
+msgstr "Ezin izan da sortu \"decodebin2\" elementua."
+
+msgid "Could not create \"queue2\" element."
+msgstr "Ezin izan da sortu \"queue2\" elementua."
+
+msgid "Could not create \"typefind\" element."
+msgstr "Ezin izan da \"typefind\" elementua sortu."
+
+#, c-format
+msgid "Error while sending data to \"%s:%d\"."
+msgstr "Errorea gertatu da datuak \"%s:%d\"(e)ra bidaltzean."
+
+#, c-format
+msgid "Error while sending gdp header data to \"%s:%d\"."
+msgstr "Errorea gertatu da gdp goiburuaren datuak \"%s:%d\"(e)ra bidaltzean."
+
+#, c-format
+msgid "Error while sending gdp payload data to \"%s:%d\"."
+msgstr ""
+"Errorea gertatu da gdp karga erabilgarriaren datuak \"%s:%d\"(e)ra "
+"bidaltzean."
+
+#, c-format
+msgid "Connection to %s:%d refused."
+msgstr "%s:%d(e)ra konektatzea ukatu da."
+
+msgid "Can't record audio fast enough"
+msgstr "Ezin da behar bezain bizkor grabatu audioa"
+
+msgid "Failed to read tag: not enough data"
+msgstr "Huts egin du etiketa irakurtzean: ez dago nahikoa daturik"
+
+msgid "track ID"
+msgstr "pistaren IDa"
+
+msgid "MusicBrainz track ID"
+msgstr "MusicBrainz pistaren IDa"
+
+msgid "artist ID"
+msgstr "interpretearen IDa"
+
+msgid "MusicBrainz artist ID"
+msgstr "MusicBrainz interpretearen IDa"
+
+msgid "album ID"
+msgstr "albumaren IDa"
+
+msgid "MusicBrainz album ID"
+msgstr "MusicBrainz albumaren IDa"
+
+msgid "album artist ID"
+msgstr "albumaren interpretearen IDa"
+
+msgid "MusicBrainz album artist ID"
+msgstr "MusicBrainz albumaren interpretearen IDa"
+
+msgid "track TRM ID"
+msgstr "pistaren TRM IDa"
+
+msgid "MusicBrainz TRM ID"
+msgstr "MusicBrainz TRM IDa"
+
+msgid "This CD has no audio tracks"
+msgstr "CD honek ez du audio-pistarik"
+
+msgid "ID3 tag"
+msgstr "ID3 etiketa"
+
+msgid "APE tag"
+msgstr "APE etiketa"
+
+msgid "ICY internet radio"
+msgstr "ICY interneteko irratia"
+
+msgid "Apple Lossless Audio (ALAC)"
+msgstr "Apple-ren galerarik gabeko audioa (ALAC)"
+
+msgid "Free Lossless Audio Codec (FLAC)"
+msgstr "Galerarik gabeko audio-kodek librea (FLAC)"
+
+msgid "Lossless True Audio (TTA)"
+msgstr "Galerarik gabeko True audioa (TTA)"
+
+msgid "Windows Media Speech"
+msgstr "Windows Media Speech"
+
+msgid "CYUV Lossless"
+msgstr "Galerarik gabeko CYUV"
+
+msgid "FFMpeg v1"
+msgstr "FFMpeg 1.b"
+
+msgid "Lossless MSZH"
+msgstr "Galerarik gabeko MSZH"
+
+msgid "Uncompressed Gray Image"
+msgstr "Konprimitu gabeko irudi grisa"
+
+msgid "Run-length encoding"
+msgstr "Run-length kodeketa"
+
+msgid "Sami subtitle format"
+msgstr "Sami azpititulu-formatua"
+
+msgid "TMPlayer subtitle format"
+msgstr "TMPlayer azpititulu-formatua"
+
+msgid "Kate subtitle format"
+msgstr "'Kate' azpititulu-formatua"
+
+msgid "Uncompressed planar YUV 4:2:0"
+msgstr "Konprimitu gabeko YUV 4:2:0 planarra"
+
+msgid "Uncompressed planar YVU 4:2:0"
+msgstr "Konprimitu gabeko YVU 4:2:0 planarra"
+
+msgid "Uncompressed packed YUV 4:2:2"
+msgstr "Konprimitu gabeko YUV 4:2:2 paketatua"
+
+msgid "Uncompressed packed YUV 4:1:0"
+msgstr "Konprimitu gabeko YUV 4:1:0 paketatua"
+
+msgid "Uncompressed packed YVU 4:1:0"
+msgstr "Konprimitu gabeko YVU 4:1:0 paketatua"
+
+msgid "Uncompressed packed YUV 4:1:1"
+msgstr "Konprimitu gabeko YUV 4:1:1 paketatua"
+
+msgid "Uncompressed packed YUV 4:4:4"
+msgstr "Konprimitu gabeko YUV 4:4:4 paketatua"
+
+msgid "Uncompressed planar YUV 4:2:2"
+msgstr "Konprimitu gabeko YUV 4:2:2 planarra"
+
+msgid "Uncompressed planar YUV 4:1:1"
+msgstr "Konprimitu gabeko YUV 4:1:1 planarra"
+
+msgid "Uncompressed black and white Y-plane"
+msgstr "Konprimitu gabeko zuri-beltzeko Y-planoa"
+
+msgid "Uncompressed YUV"
+msgstr "Konprimitu gabeko YUV"
+
+#, c-format
+msgid "Uncompressed palettized %d-bit %s"
+msgstr "Konprimitu gabeko paleta moduko %d biteko %s"
+
+#, c-format
+msgid "DivX MPEG-4 Version %d"
+msgstr "DivX MPEG-4 %d. bertsioa"
+
+#, c-format
+msgid "Raw %d-bit PCM audio"
+msgstr "%d biteko prozesatu gabeko PCM audioa"
+
+msgid "Raw PCM audio"
+msgstr "Prozesatu gabeko PCM audioa"
+
+#, c-format
+msgid "Raw %d-bit floating-point audio"
+msgstr "Koma mugikorreko %d biteko prozesatu gabeko PCM audioa"
+
+msgid "Raw floating-point audio"
+msgstr "Koma mugikorreko prozesatu gabeko audioa"
+
+msgid "Audio CD source"
+msgstr "Audio CDaren iturburua"
+
+msgid "DVD source"
+msgstr "DVD iturburua"
+
+msgid "Real Time Streaming Protocol (RTSP) source"
+msgstr "Denbora errealeko transmisio-protokoloaren (RTSP) iturburua"
+
+msgid "Microsoft Media Server (MMS) protocol source"
+msgstr "Microsoft Media Server (MMS) protokoloaren iturburua"
+
+#, c-format
+msgid "%s protocol source"
+msgstr "%s protokolo-ituruburua"
+
+#, c-format
+msgid "%s video RTP depayloader"
+msgstr "%s bideoaren RTP depayloader-a"
+
+#, c-format
+msgid "%s audio RTP depayloader"
+msgstr "%s audiooaren RTP depayloader-a"
+
+#, c-format
+msgid "%s RTP depayloader"
+msgstr "%s RTP depayloader-a"
+
+#, c-format
+msgid "%s demuxer"
+msgstr "%s demultiplexadorea"
+
+#, c-format
+msgid "%s decoder"
+msgstr "%s deskodetzailea"
+
+#, c-format
+msgid "%s video RTP payloader"
+msgstr "%s bideoaren RTP payloader-a"
+
+#, c-format
+msgid "%s audio RTP payloader"
+msgstr "%s audiooaren RTP payloader-a"
+
+#, c-format
+msgid "%s RTP payloader"
+msgstr "%s RTP payloader-a"
+
+#, c-format
+msgid "%s muxer"
+msgstr "%s multiplexadorea"
+
+#, c-format
+msgid "%s encoder"
+msgstr "%s kodetzailea"
+
+#, c-format
+msgid "GStreamer element %s"
+msgstr "%s GStreamer elementua"
+
+msgid "Unknown source element"
+msgstr "Iturburu-elementu ezezaguna"
+
+msgid "Unknown sink element"
+msgstr "Kolektore-elementu ezezaguna"
+
+msgid "Unknown element"
+msgstr "Elementu ezezaguna"
+
+msgid "Unknown decoder element"
+msgstr "Deskodetzailearen elementu ezezaguna"
+
+msgid "Unknown encoder element"
+msgstr "Kodetzailearen elementu ezezaguna"
+
+msgid "Plugin or element of unknown type"
+msgstr "Mota ezezaguneko plugina edo elementua"
+
+msgid "No device specified."
+msgstr "Ez da gailurik zehaztu."
+
+#, c-format
+msgid "Device \"%s\" does not exist."
+msgstr "\"%s\" gailua ez dago."
+
+#, c-format
+msgid "Device \"%s\" is already being used."
+msgstr "\"%s\" gailua erabiltzen ari dira."
+
+#, c-format
+msgid "Could not open device \"%s\" for reading and writing."
+msgstr "Ezin izan da \"%s\" gailua ireki irakurtzeko eta idazteko."
+
+#, fuzzy
+#~ msgid "No file name specified."
+#~ msgstr "Ez da gailurik zehaztu."
projects / platform / upstream / gstreamer.git / commitdiff