gst-plugins-base 0.10.25
- From: "Jan Schmidt" <install-module gnome org>
- To: FTP Releases <ftp-release-list gnome org>
- Subject: gst-plugins-base 0.10.25
- Date: Mon, 5 Oct 2009 15:09:11 +0000 (UTC)
Module: gst-plugins-base
Version: 0.10.25
Uploaded by: Jan Schmidt
http://download.gnome.org/sources/gst-plugins-base/0.10/gst-plugins-base-0.10.25.tar.gz
sha256sum: a37226aba3a0c95d02a2a15a5dd7af53063787ada88076c84a2ee6f18e8bbdf4
size: 3.1M
http://download.gnome.org/sources/gst-plugins-base/0.10/gst-plugins-base-0.10.25.tar.bz2
sha256sum: 0ab2f7e1d818e7af1be99c4eae02ba69d4a1b8f7e3527929a6426f1daa0d4607
size: 2.2M
News
----
This is GStreamer Base Plug-ins 0.10.25, "Standard disclaimers apply"
Changes since 0.10.24:
* Add per-stream volume controls
* Theora 1.0 and Y444 and Y42B format support
* Improve audio capture timing
* GObject introspection support
* Improve audio output startup
* RTSP improvements
* Use pango-cairo instead of pangoft2
* Allow cdda://(device#)?track URI scheme in cddabasesrc
* Support interlaced content in videoscale and ffmpegcolorspacee
* Many other bug fixes and improvements
Bugs fixed since 0.10.24:
* 595401 : gobject assertion and null access to volume instance in playbin
* 563828 : [decodebin2] Complains about loops in the graph when demuxer output requires another demuxer
* 591677 : Easy codec installation is not working
* 588523 : smarter sink selection in playbin2
* 590146 : adder regressions
* 321532 : [cddabasesrc] Support device setting in cdda:// URI
* 340887 : add pangocairo textoverlay plugin.
* 397419 : [oggdemux] ogm video with subtitles stuck on first frame
* 556537 : [PATCH] typefind: more flexible MPEG4 start code recognition
* 559049 : gstcheck.c:76:F:general:test_state_changes_* failure: GST_IS_CLOCK(clock) assertion fails
* 567660 : [API] need a stream volume interface for sinks that do volume control
* 567928 : Make videorate work with a live source
* 571610 : [playbin] Scale of volume property is not documented
* 583255 : [playbin2] deadlock when disabling visualisations
* 586180 : RTSP improvements
* 588717 : [oggmux] gst_caps_unref() warning if not linked downstream
* 588761 : [videoscale] Needs special support for interlaced content
* 588915 : audioresample's output offset counter's initialization could maybe be improved
* 589095 : [appsrc] clarify documentation on caps and linkage
* 589574 : [typefind] incorrect sdp file detection
* 590243 : [videoscale] Claims to support MAX width/height
* 590425 : Slaved alsasrc clock with slave-method=re-timestamp not usable for RTP audio
* 590856 : [decodebin2] triggers assertion failure on NULL caps
* 591207 : totem does display the following subtitle srt file.
* 591357 : gst-plugins-base git won't build due to warning in gstrtspconnection.c
* 591577 : [playbin2] Incorrect error message string
* 591664 : [playbin2] after seeking, srt subtitles don't resync correctly
* 591934 : timestamp drift in audioresample
* 592544 : Remove regex.h check
* 592657 : [appsink] Blocks after entering on pause state
* 592864 : deadlocks from recent inputselector/streamselector change
* 592884 : [playbin2] g_object_get increases refcount by 2 and therefore leaves memleak
* 593035 : gdp doesn't preserve fields of the buffers put into the caps' streamheader
* 593284 : basertppayloader takes time in instance init
* 594020 : Totem don't play videos from ssh remote host
* 594094 : Playback Error playing Midi file
* 594136 : [alsasink] Regression from 0.10.23 -- element reuse doesn't work
* 594165 : [theoraenc] Implement support for new formats
* 594256 : improved slave-skew resynch mechanism
* 594258 : missing break in rtcpbuffer
* 594275 : Add cast to navigation to fix compiler warning
* 594623 : Expose playsink as a fully-fledged element
* 594732 : parse error
* 594757 : build fails due to warning in gstbasertppayload.c
* 594993 : [introspection] pkg-config file madness
* 594994 : [streamvolume] Add get_type function to the documentation
* 595454 : [cddabasesrc] uri format change breaks rhythmbox
* 545807 : [baseaudiosink] audible crack when starting the pipeline
API added since 0.10.24:
* gst_rtsp_connection_create_from_fd()
* gst_rtsp_connection_set_http_mode()
* gst_rtsp_watch_write_data()
* gst_rtsp_watch_send_message()
* GstBaseRTPPayload::perfect-rtptime
* GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
* GstVideoSinkClass::show_frame()
* GstVideoSink:show-preroll-frame
* GST_MIXER_TRACK_READONLY
* GST_MIXER_TRACK_WRITEONLY
* GstStreamVolume interface
--
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