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Release 0.10.14
Original commit message from CVS:
Release 0.10.14
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Jan Schmidt committed Aug 3, 2007
1 parent 42771c4 commit 221ae4e
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7 changes: 7 additions & 0 deletions ChangeLog
@@ -1,3 +1,10 @@
=== release 0.10.14 ===

2007-08-03 Jan Schmidt <thaytan@mad.scientist.com>

* configure.ac:
releasing 0.10.14, "Light Years Ahead"

2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>

* tests/check/libs/audio.c: (GST_START_TEST):
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55 changes: 54 additions & 1 deletion NEWS
@@ -1,10 +1,63 @@
This is GStreamer Base Plug-ins 0.10.13, "What's going on?"
This is GStreamer Base Plug-ins 0.10.14, "Light Years Ahead"

Please note that decodebin2 API included in this release is still
considered unstable and WILL change in future releases. At this stage, only
developers or early adopters should consider using the decodebin2 API embodied
in its signals and properties.

Changes since 0.10.13:

* Audio dither and noise-shaping when reducing bit-depth
* RTSP and SDP helper libraries added
* Experimental buffering element "queue2" now supports pull-mode
and file-based buffering.
* Support for more 32-bit video pixel layouts
* Various fixes and improvements

Bugs fixed since 0.10.13:

* 380625 : [x*imagesink] add 'handle-expose' property
* 385527 : oggmux sometimes gets DELTA flag on output wrong near start
* 402076 : videoscale 4-tap method broken for downscaling
* 437169 : [xvimagesink] add property to disable Xv double-buffering
* 441264 : queue2 support to do buffering on a file
* 442553 : [v4lsrc] doesn't output segments in GST_FORMAT_TIME
* 442557 : [videorate] doesn't handle latency queries
* 442944 : Audiotestsrc can overflow on seeks
* 444523 : [queue2] Pull mode support
* 444630 : Compilation error with fsseko (from gstqueue2.c) -- unabl...
* 445505 : [queue2] It does not work in pull mode with oggdemux
* 446551 : [queue2] Buffering is not working properly if it is set t...
* 446572 : [queue2] Division by zero
* 446972 : warning when compiling gstoggdemux.c
* 449156 : Regression in CVS for decodebin2
* 450875 : Missing files in po/POTFILES.in
* 451707 : [tag] UTF-8 in ID3v1 tag not correctly decoded
* 451908 : [ffmpegcolorspace] regression: doesn't accept GST_VIDEO_C...
* 454264 : Playbin fails to " play " image url after a movie url
* 456656 : [API] Addition of audio buffer clipping function to gstaudio
* 460978 : gst_audio_buffer_clip outputs warnings
* 152864 : [PATCH] GstAlsaMixer doesn't support signals
* 360246 : [audioconvert] Optionally apply dithering
* 394061 : Add support for Subviewer subtitles
* 420326 : Base payloader class has wrong property types and ranges
* 451145 : [vorbisdec] errors out on 0-sized packets
* 459204 : [PATCH] [playbin] gst_play_base_bin_get_streaminfo_value_...

API added since 0.10.13:

* RTSP and SDP libraries added
* gst_rtsp_base64_decode_ip
* Add buffer clipping function gst_audio_buffer_clip for raw audio
buffers. Fixes #456656.
* gst_mixer_get_mixer_flags
* gst_mixer_message_parse_mute_toggled
* gst_mixer_message_parse_record_toggled
* gst_mixer_message_parse_volume_changed
* gst_mixer_message_parse_option_changed
* GstMixerMessageType
* GstMixerFlags

Changes since 0.10.12:
* Many fixes and improvements
* RTP and RTCP support improved
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98 changes: 47 additions & 51 deletions RELEASE
@@ -1,5 +1,5 @@

Release notes for GStreamer Base Plug-ins 0.10.13 "What's Going on?"
Release notes for GStreamer Base Plug-ins 0.10.14 "Light Years Ahead"



Expand Down Expand Up @@ -54,53 +54,59 @@ contains a set of less supported plug-ins that haven't passed the

Features of this release

* Many fixes and improvements
* RTP and RTCP support improved
* Audio dither and noise-shaping when reducing bit-depth
* RTSP and SDP helper libraries added
* Experimental buffering element "queue2" now supports pull-mode
and file-based buffering.
* Support for more 32-bit video pixel layouts
* Various fixes and improvements
* Parallel installability with 0.8.x series
* Threadsafe design and API

Bugs fixed in this release

* 339838 : [audioconvert] support floats with non-native endianness
* 393975 : closing x/xvimagesink window crashes gst-launch
* 405072 : [API] add gst_tag_freeform_string_to_utf8()
* 413799 : [subparse] add support for MPL2 format
* 414645 : GstMixerTrack should make untranslated label available
* 420079 : [audioconvert] Uses biased rounding which results in dist...
* 420578 : [subparse] add more colour map in sami parser
* 421834 : videorate breaks on dimension changes
* 423051 : Vorbis tags of type double use locale-dependent formatting
* 423055 : Verify ReplayGain vorbistag processing in libs/tag testsuite
* 425455 : Decodebin2 leaks pads
* 426250 : GstPlayBaseBin leaks streaminfo objects
* 428187 : Rtp base depayloader class doesn't send new_segment after...
* 431672 : gst_base_rtp_audio_payload_push() should take object of i...
* 432362 : [ximagesink] doesn't build if XShm is not available
* 432755 : [videorate] leaks buffer if flow != OK
* 432984 : [baseaudiosrc] misleading warning message when dropping s...
* 433888 : [theoradec] does not generate a perfect stream
* 436562 : Theoradec doesn't work well with gnonlin
* 438840 : [theoradec] does not compile with old version of libtheora
* 440997 : [gstriff] Doesn't handle width!=depth files with audio/x-...
* 441295 : audioconvert doesn't build on VS6
* 442024 : regression in playbin buffering
* 350299 : [playbin] " Internal data flow error " opening movie with s...
* 410039 : totem crashed with SIGSEGV in new_decoded_pad_full()
* 340842 : do latency calculation for live sources
* 341078 : RB does not play beyond initially downloaded podcast file
* 414496 : [id3demux, id3v2mux] Add support for GST_TAG_MUSICBRAINZ_...
* 380625 : [x*imagesink] add 'handle-expose' property
* 385527 : oggmux sometimes gets DELTA flag on output wrong near start
* 402076 : videoscale 4-tap method broken for downscaling
* 437169 : [xvimagesink] add property to disable Xv double-buffering
* 441264 : queue2 support to do buffering on a file
* 442553 : [v4lsrc] doesn't output segments in GST_FORMAT_TIME
* 442557 : [videorate] doesn't handle latency queries
* 442944 : Audiotestsrc can overflow on seeks
* 444523 : [queue2] Pull mode support
* 444630 : Compilation error with fsseko (from gstqueue2.c) -- unabl...
* 445505 : [queue2] It does not work in pull mode with oggdemux
* 446551 : [queue2] Buffering is not working properly if it is set t...
* 446572 : [queue2] Division by zero
* 446972 : warning when compiling gstoggdemux.c
* 449156 : Regression in CVS for decodebin2
* 450875 : Missing files in po/POTFILES.in
* 451707 : [tag] UTF-8 in ID3v1 tag not correctly decoded
* 451908 : [ffmpegcolorspace] regression: doesn't accept GST_VIDEO_C...
* 454264 : Playbin fails to " play " image url after a movie url
* 456656 : [API] Addition of audio buffer clipping function to gstaudio
* 460978 : gst_audio_buffer_clip outputs warnings
* 152864 : [PATCH] GstAlsaMixer doesn't support signals
* 360246 : [audioconvert] Optionally apply dithering
* 394061 : Add support for Subviewer subtitles
* 420326 : Base payloader class has wrong property types and ranges
* 451145 : [vorbisdec] errors out on 0-sized packets
* 459204 : [PATCH] [playbin] gst_play_base_bin_get_streaminfo_value_...

API changed in this release

- API additions:

* add gst_tag_freeform_string_to_utf8()
* GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
* GstBaseAudioSink::slave-method property
* add "min-ptime" property to RTP base audio payloader
* gst_base_rtp_audio_payload_push()
* gst_base_rtp_audio_payload_get_adapter()
* GstMixerTrack::untranslated-label property
* RTSP and SDP libraries added
* gst_rtsp_base64_decode_ip
* Add buffer clipping function gst_audio_buffer_clip for raw audio buffers. Fixes #456656.
* gst_mixer_get_mixer_flags
* gst_mixer_message_parse_mute_toggled
* gst_mixer_message_parse_record_toggled
* gst_mixer_message_parse_volume_changed
* gst_mixer_message_parse_option_changed
* GstMixerMessageType
* GstMixerFlags

Download

Expand Down Expand Up @@ -130,29 +136,19 @@ Applications

Contributors to this release

* Alex Lancaster
* Andy Wingo
* Christian Kirbach
* Bastien Nocera
* Dan Williams
* David Schleef
* Edward Hervey
* Jan Schmidt
* Julien MOUTTE
* Kamil Pawlowski
* Marc-Andre Lureau
* Mark Nauwelaerts
* Jorn Baayen
* Michael Smith
* Olivier Crete
* René Stadler
* Sebastian Dröge
* Sebastien Moutte
* Stefan Kost
* Thiago Sousa Santos
* Thomas Vander Stichele
* Tim-Philipp Müller
* Tommi Myöhänen
* Vincent Torri
* Wim Taymans
* Young-Ho Cha
* Zaheer Abbas Merali
* Zeeshan Ali

4 changes: 2 additions & 2 deletions configure.ac
Expand Up @@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
dnl initialize autoconf
dnl releases only do -Wall, cvs and prerelease does -Werror too
dnl use a three digit version number for releases, and four for cvs/prerelease
AC_INIT(GStreamer Base Plug-ins, 0.10.13.1,
AC_INIT(GStreamer Base Plug-ins, 0.10.14,
http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer,
gst-plugins-base)

Expand Down Expand Up @@ -44,7 +44,7 @@ dnl - interfaces added/removed/changed -> increment CURRENT, REVISION = 0
dnl - interfaces added -> increment AGE
dnl - interfaces removed -> AGE = 0
dnl sets GST_LT_LDFLAGS
AS_LIBTOOL(GST, 9, 0, 9)
AS_LIBTOOL(GST, 10, 0, 10)

dnl FIXME: this macro doesn't actually work;
dnl the generated libtool script has no support for the listed tags.
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