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author | Jan Schmidt <thaytan@mad.scientist.com> | 2008-01-28 23:31:26 +0000 |
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committer | Jan Schmidt <thaytan@mad.scientist.com> | 2008-01-28 23:31:26 +0000 |
commit | 018ab2ef4c2860350549ab874bab45a0fd052ee8 (patch) | |
tree | 3323789dae87fcc6453b01c60597c6b8ed348d29 /NEWS | |
parent | ac6402a0daf83fe0491a2bf11bc336a59ea1d183 (diff) | |
download | gstreamer-plugins-base-018ab2ef4c2860350549ab874bab45a0fd052ee8.tar.gz |
Release 0.10.16RELEASE-0_10_16
Original commit message from CVS:
Release 0.10.16
Diffstat (limited to 'NEWS')
-rw-r--r-- | NEWS | 72 |
1 files changed, 67 insertions, 5 deletions
@@ -1,9 +1,71 @@ -This is GStreamer Base Plug-ins 0.10.15, "No need to argue" +This is GStreamer Base Plug-ins 0.10.16, "Scheduled Interruption" -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. +IMPORTANT NOTES + +1) Please note that decodebin2 and playbin2 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 decodebin2 or playbin2 +API embodied in their signals and properties. + +2) On some systems, the current release of gst-plugins-good (0.10.6) may fail to +build against this release of gst-plugins-base with an error like: + +gstid3v2mux.cc:547: error: 'GST_TAG_MUSICBRAINZ_SORTNAME' was not declared in this scope + +In this case, you should either patch the configure file of gst-plugins-good to +remove -DGST_DISABLE_DEPRECATED from DEPRECATED_CFLAGS=, or else compile +with make DEPRECATED_CFLAGS='' + +3) Some users may experience problems using the 'mp3parse' element from the +previous gst-plugins-ugly release (0.10.6). This is due to a bug in mp3parse +exposed by changes in decodebin in gst-plugins-base. It will be fixed in the +upcoming release of gst-plugins-ugly next month. In the meantime as a +workaround, you can set the rank of mp3parse to GST_RANK_NONE in +gst-plugins-ugly/gst/mpegaudioparse/gstmpegaudioparse.c when compiling, or +or remove the /usr/lib/gstreamer-0.10/libgstmpegaudioparse.so file entirely. + +Changes since 0.10.15: + + * Handle newer Theora granule-pos semantics + * Introducing first alpha version playbin2 - the upcoming successor to + playbin + * Fixes in playbin handling of stream-switching + * New API for uniform handling of raw-video format buffers. + * Improvements for RTSP/RTP handling + * RIFF lib additions for VC-1 and AVC1 fourccs + * Many other bug-fixes and improvements + +Bugs fixed since 0.10.15: + + * 506132 : Review of changes in video/video.h + * 320984 : [oggdemux] cannot handle multiple chains + * 373011 : [playbin] throws error when switching off subtitles + * 436756 : Intermittent crashes in Pidgin in audioclock g_type_class... + * 462740 : [streamselector] patch to improve default stream selection + * 486840 : [alsamixer] use _all variants when setting the mixer + * 497964 : theoraenc test fails + * 498228 : gst-plugins-base-0.10.15 does not compile on FreeBSD (Gen... + * 499697 : Provide better pkg-config files + * 502497 : [subparse] SubRip subtitles starting from 0 not recognised + * 503440 : The control sockets used by gstrtspconnection.c are never... + * 503930 : [cdda] warning: 'eos' may be used uninitialized in this f... + * 506928 : [alsamixer] add " PCM " as master fall back for cards that ... + * 508138 : [decodebin] does not error out if pad activation fails + * 509762 : missing file in win32/MANIFEST + * 511274 : gst_rtp_buffer_get_extension_data is returning FALSE when... + * 496731 : [PATCH] xvimagesink leaks memory if initialization fails + * 496761 : [PATCH] RTSP message leaks memory when uninitialized + * 500763 : SIGSEGV while playing ogg audio file + +API additions since 0.10.15: + + * New GstVideoFormat API and helper functions in libgstvideo + * gst_base_audio_sink_set_provide_clock() + * gst_base_audio_sink_get_provide_clock() + * gst_base_audio_sink_set_slave_method() + * gst_base_audio_sink_get_slave_method() + * gst_base_audio_src_set_provide_clock() + * gst_base_audio_src_get_provide_clock() Changes since 0.10.14: |