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authorJan Schmidt <thaytan@mad.scientist.com>2007-11-16 00:14:33 +0000
committerJan Schmidt <thaytan@mad.scientist.com>2007-11-16 00:14:33 +0000
commit15be4ee905aac46ca067d45db4b4f9c9fe5f02be (patch)
tree692477f71a38420c0a81952d18e0839e84016b47 /RELEASE
parent5424e697fb5ae8b0874543f3b75c810b3d0a779e (diff)
downloadgstreamer-plugins-base-15be4ee905aac46ca067d45db4b4f9c9fe5f02be.tar.gz
configure.ac: releasing 0.10.15, "No need to argue"RELEASE-0_10_15
Original commit message from CVS: === release 0.10.15 === 2007-11-15 Jan Schmidt <jan.schmidt@sun.com> * configure.ac: releasing 0.10.15, "No need to argue"
Diffstat (limited to 'RELEASE')
-rw-r--r--RELEASE153
1 files changed, 102 insertions, 51 deletions
diff --git a/RELEASE b/RELEASE
index 3637eeb79..02fed9782 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,5 +1,5 @@
-Release notes for GStreamer Base Plug-ins 0.10.14 "Light Years Ahead"
+Release notes for GStreamer Base Plug-ins 0.10.15 "No need to argue"
@@ -54,59 +54,89 @@ contains a set of less supported plug-ins that haven't passed the
Features of this release
- * 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
+ * RTP/RTSP/RTCP/SDP support improved
+ * New FFT support library libgstfft, based on Kiss FFT
+ * New formats supported in volume and audiotestsrc
+ * Fixes in audiorate and videorate
+ * Audio capture fixes
+ * Playbin and decodebin fixes
+ * New tagdemux base class for ID3/APE style tag readers
+ * Fix a nasty crash in the X sinks on shutdown
+ * New tags supported
+ * Add support for multichannel WAV files.
+ * Preserve channel layout information when up/down-mixing.
+ * Many bug-fixes and improvements
+ *
Bugs fixed in this release
- * 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_...
+ * 475395 : decodebin2 leaks request-pads
+ * 475451 : [decodebin2] leaks ghostpad
+ * 378770 : [xvimagesink] race condition in event thread?
+ * 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
+ * 430677 : [audioconvert] does not preserve channel positions when f...
+ * 442654 : [volume] controller bypassed by default
+ * 445529 : [volume] support for 24/32-bit audio/x-raw-int
+ * 446766 : return code for gst_base_rtp_payload_audio_handle_event()
+ * 451970 : Subparse requires HTML parser
+ * 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
+ * 459334 : [textoverlay] expose pango line alignment property
+ * 459585 : [basertpdepayload] api without namespace
+ * 460422 : [audiotestsrc] Add support for float and double output
+ * 462805 : [alsa] compilation fails with gcc 4.2
+ * 462979 : Add 'silent' property to GstTimeOverlay
+ * 463215 : [audioconvert] compile errors
+ * 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
+ * 464666 : [playbin] QT trailer hangs in preroll with decodebin2
+ * 464690 : Add connection-speed property to uridecodebin element
+ * 465015 : [playbin] Not removed probes causes deadlocks in streamin...
+ * 465028 : some warnings with mingw
+ * 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
+ * 468129 : [basertpaudiopayload] event handler returns the wrong value
+ * 468619 : New library gstfft: FFT library for integer and float typ...
+ * 470456 : [API] add gst_missing_*_installer_detail_new()
+ * 470766 : [ssaparse] line breaks in SSA subtitle parser
+ * 471067 : Make the SDP code useable for generating SDP descriptions
+ * 471194 : [rtpbuffer] RTP headers are wrong for win32
+ * 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
+ * 474384 : gstrtsp-enumtypes.c and .h needed for win32
+ * 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
+ * 475731 : rtspconnection is able to read incomplete messages
+ * 483620 : All Rtp buffers are discarded -- gst_rtp_buffer_get_payl...
+ * 484989 : memleak, not unrefed caps for gstbasertppayload.c
+ * 489010 : Please change default channel order for WAVE_EXT-less .wa...
+ * 491722 : [playbin] regression: crash with external subtitles
+ * 492098 : [GstFFT] Broken scaling
+ * 492114 : Build issues on Windows/MSVC
+ * 492306 : compilation errors with MinGW
+ * 492813 : Missing symbols in libgstrtp.def
+ * 493986 : Build issues on Windows (missing symbols)
+ * 494346 : pre-release vs6 patch
+ * 496548 : Including malloc.h breaks macos build
+ * 496724 : DSW file references non-existent DSP files
+ * 464079 : audiotestsrc doesn't respond to conversion queries properly
+ * 442065 : floatcast.h includes config.h and might break other apps
+ * 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
+ * 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
+ * 464028 : Move connection-speed from playbin to playbasebin
API changed in this release
- API additions:
-* 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
+* GstTagDemux base class for simple tag demuxers
+* GstBaseAudioSrc::provide-clock property
+* gst_rtcp_ntp_to_unix()
+* gst_rtcp_unix_to_ntp()
+* gst_rtp_buffer_get_header_len()
+* gst_rtp_buffer_get_extension_data()
+* gst_rtp_buffer_compare_seqnum()
+* gst_rtp_buffer_ext_timestamp()
+* gst_rtcp_packet_sdes_copy_entry()
+* gst_install_plugins_supported()
+* gst_missing_*_installer_detail_new() convenience API
+* gst_rtsp_connection_poll()
+* GstTextOverlay::line-alignment property
Download
@@ -136,19 +166,40 @@ Applications
Contributors to this release
- * Andy Wingo
- * Bastien Nocera
+ * Stefan Kost
+ * Alexander Shopov
+ * Damien Lespiau
* Dan Williams
+ * Daniel Díaz
* David Schleef
- * Edward Hervey
+ * Davyd Madeley
+ * Funda Wang
+ * Haakon Sporsheim
+ * Ilkka Tuohela
+ * Jakub Bogusz
* Jan Schmidt
- * Jorn Baayen
+ * Jason Kivlighn
+ * Jens Granseuer
+ * Johan Dahlin
+ * Jorge González González
+ * Josep Torra Valles
+ * Julien MOUTTE
+ * Laurent Glayal
* Michael Smith
+ * Mogens Jaeger
+ * Ole André Vadla Ravnås
+ * Olivier Crete
+ * Peter Kjellerstedt
+ * Renato Filho
+ * René Stadler
* Sebastian Dröge
* Sebastien Moutte
* Stefan Kost
- * Thiago Sousa Santos
+ * Thijs Vermeir
* Thomas Vander Stichele
* Tim-Philipp Müller
+ * Tommi Myöhänen
+ * Vincent Torri
* Wim Taymans
+ * Yang Hong
  \ No newline at end of file