diff options
author | Jan Schmidt <thaytan@mad.scientist.com> | 2007-11-16 00:14:33 +0000 |
---|---|---|
committer | Jan Schmidt <thaytan@mad.scientist.com> | 2007-11-16 00:14:33 +0000 |
commit | 15be4ee905aac46ca067d45db4b4f9c9fe5f02be (patch) | |
tree | 692477f71a38420c0a81952d18e0839e84016b47 /RELEASE | |
parent | 5424e697fb5ae8b0874543f3b75c810b3d0a779e (diff) |
configure.ac: releasing 0.10.15, "No need to argue"
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-- | RELEASE | 153 |
1 files changed, 102 insertions, 51 deletions
@@ -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
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