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<?xml version="1.0"?>
<?xml-stylesheet href="../releases.xsl" type="text/xsl"?>
<release>
<module>gst-plugins-base</module>
<module-fancy>GStreamer Base Plug-ins</module-fancy>
<name>Standard disclaimers apply</name>
<version>0.10.25</version>
<intro>
<p>
The GStreamer team is proud to announce a new release
in the 0.10.x stable series of the
GStreamer Base Plug-ins.
</p>
<p>
The 0.10.x series is a stable series targeted at end users.
It is not API or ABI compatible with the stable 0.8.x series.
It is, however, parallel installable with the 0.8.x series.
</p>
<p>
This module contains a set of reference plugins, base classes for other
plugins, and helper libraries.
This module is kept up-to-date together with the core developments. Element
writers should look at the elements in this module as a reference for
their development.
This module contains elements for, among others:
<ul>
<li>device plugins: x(v)imagesink, alsa, v4lsrc, cdparanoia</li>
<li>containers: ogg</li>
<li>codecs: vorbis, theora</li>
<li>text: textoverlay, subparse</li>
<li>sources: audiotestsrc, videotestsrc, gnomevfssrc</li>
<li>network: tcp</li>
<li>typefind</li>
<li>audio processing: audioconvert, adder, audiorate, audioscale, volume</li>
<li>visualisation: libvisual</li>
<li>video processing: ffmpegcolorspace</li>
<li>aggregate elements: decodebin, playbin</li>
</ul>
Other modules containing plug-ins are:
<dl>
<dt>gst-plugins-good</dt>
<dd>contains a set of well-supported plug-ins under our preferred license</dd>
<dt>gst-plugins-ugly</dt>
<dd>contains a set of well-supported plug-ins, but might pose problems for
distributors</dd>
<dt>gst-plugins-bad</dt>
<dd>contains a set of less supported plug-ins that haven't passed the
rigorous quality testing we expect</dd>
</dl>
</p>
</intro>
<features>
<feature>Add per-stream volume controls</feature>
<feature>Theora 1.0 and Y444 and Y42B format support</feature>
<feature>Improve audio capture timing</feature>
<feature>GObject introspection support</feature>
<feature>Improve audio output startup</feature>
<feature>RTSP improvements</feature>
<feature>Use pango-cairo instead of pangoft2</feature>
<feature>Allow cdda://(device#)?track URI scheme in cddabasesrc</feature>
<feature>Support interlaced content in videoscale and ffmpegcolorspacee</feature>
<feature>Many other bug fixes and improvements</feature>
</features>
<applications>
</applications>
<contributors>
<person>Arnout Vandecappelle</person>
<person>Benjamin Gaignard</person>
<person>Benjamin Otte</person>
<person>Christian F.K. Schaller</person>
<person>David Schleef</person>
<person>Edward Hervey</person>
<person>Eero Nurkkala</person>
<person>Havard Graff</person>
<person>Håvard Graff</person>
<person>Jan Schmidt</person>
<person>John Millikin</person>
<person>Jonas Holmberg</person>
<person>Jonathan Matthew</person>
<person>Josep Torra</person>
<person>Kipp Cannon</person>
<person>Marc-André Lureau</person>
<person>Mark Nauwelaerts</person>
<person>Mart Raudsepp</person>
<person>Michael Smith</person>
<person>Olivier Crête</person>
<person>Peter Kjellerstedt</person>
<person>Philip Jägenstedt</person>
<person>René Stadler</person>
<person>Sebastian Dröge</person>
<person>Siarhei Siamashka</person>
<person>Stefan Kost</person>
<person>Tim-Philipp Müller</person>
<person>Wim Taymans</person>
<person>Young-Ho Cha</person>
<person>Руслан Ижбулатов</person>
</contributors>
<api>
<additions>
<item>gst_rtsp_connection_create_from_fd()</item>
<item>gst_rtsp_connection_set_http_mode()</item>
<item>gst_rtsp_watch_write_data()</item>
<item>gst_rtsp_watch_send_message()</item>
<item>GstBaseRTPPayload::perfect-rtptime</item>
<item>GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()</item>
<item>GstVideoSinkClass::show_frame()</item>
<item>GstVideoSink:show-preroll-frame</item>
<item>GST_MIXER_TRACK_READONLY</item>
<item>GST_MIXER_TRACK_WRITEONLY</item>
<item>GstStreamVolume interface</item>
</additions>
<!--
<deprecations>
</deprecations>
-->
</api>
<bugs>
<bug>
<id>595401</id>
<summary>gobject assertion and null access to volume instance in playbin</summary>
</bug>
<bug>
<id>563828</id>
<summary>[decodebin2] Complains about loops in the graph when demuxer output requires another demuxer</summary>
</bug>
<bug>
<id>591677</id>
<summary>Easy codec installation is not working</summary>
</bug>
<bug>
<id>588523</id>
<summary>smarter sink selection in playbin2</summary>
</bug>
<bug>
<id>590146</id>
<summary>adder regressions</summary>
</bug>
<bug>
<id>321532</id>
<summary>[cddabasesrc] Support device setting in cdda:// URI</summary>
</bug>
<bug>
<id>340887</id>
<summary>add pangocairo textoverlay plugin.</summary>
</bug>
<bug>
<id>397419</id>
<summary>[oggdemux] ogm video with subtitles stuck on first frame</summary>
</bug>
<bug>
<id>556537</id>
<summary>[PATCH] typefind: more flexible MPEG4 start code recognition</summary>
</bug>
<bug>
<id>559049</id>
<summary>gstcheck.c:76:F:general:test_state_changes_* failure: GST_IS_CLOCK(clock) assertion fails</summary>
</bug>
<bug>
<id>567660</id>
<summary>[API] need a stream volume interface for sinks that do volume control</summary>
</bug>
<bug>
<id>567928</id>
<summary>Make videorate work with a live source</summary>
</bug>
<bug>
<id>571610</id>
<summary>[playbin] Scale of volume property is not documented</summary>
</bug>
<bug>
<id>583255</id>
<summary>[playbin2] deadlock when disabling visualisations</summary>
</bug>
<bug>
<id>586180</id>
<summary>RTSP improvements</summary>
</bug>
<bug>
<id>588717</id>
<summary>[oggmux] gst_caps_unref() warning if not linked downstream</summary>
</bug>
<bug>
<id>588761</id>
<summary>[videoscale] Needs special support for interlaced content</summary>
</bug>
<bug>
<id>588915</id>
<summary>audioresample's output offset counter's initialization could maybe be improved</summary>
</bug>
<bug>
<id>589095</id>
<summary>[appsrc] clarify documentation on caps and linkage</summary>
</bug>
<bug>
<id>589574</id>
<summary>[typefind] incorrect sdp file detection</summary>
</bug>
<bug>
<id>590243</id>
<summary>[videoscale] Claims to support MAX width/height</summary>
</bug>
<bug>
<id>590425</id>
<summary>Slaved alsasrc clock with slave-method=re-timestamp not usable for RTP audio</summary>
</bug>
<bug>
<id>590856</id>
<summary>[decodebin2] triggers assertion failure on NULL caps</summary>
</bug>
<bug>
<id>591207</id>
<summary>totem does display the following subtitle srt file.</summary>
</bug>
<bug>
<id>591357</id>
<summary>gst-plugins-base git won't build due to warning in gstrtspconnection.c</summary>
</bug>
<bug>
<id>591577</id>
<summary>[playbin2] Incorrect error message string</summary>
</bug>
<bug>
<id>591664</id>
<summary>[playbin2] after seeking, srt subtitles don't resync correctly</summary>
</bug>
<bug>
<id>591934</id>
<summary>timestamp drift in audioresample</summary>
</bug>
<bug>
<id>592544</id>
<summary>Remove regex.h check</summary>
</bug>
<bug>
<id>592657</id>
<summary>[appsink] Blocks after entering on pause state</summary>
</bug>
<bug>
<id>592864</id>
<summary>deadlocks from recent inputselector/streamselector change</summary>
</bug>
<bug>
<id>592884</id>
<summary>[playbin2] g_object_get increases refcount by 2 and therefore leaves memleak</summary>
</bug>
<bug>
<id>593035</id>
<summary>gdp doesn't preserve fields of the buffers put into the caps' streamheader</summary>
</bug>
<bug>
<id>593284</id>
<summary>basertppayloader takes time in instance init</summary>
</bug>
<bug>
<id>594020</id>
<summary>Totem don't play videos from ssh remote host</summary>
</bug>
<bug>
<id>594094</id>
<summary>Playback Error playing Midi file</summary>
</bug>
<bug>
<id>594136</id>
<summary>[alsasink] Regression from 0.10.23 -- element reuse doesn't work</summary>
</bug>
<bug>
<id>594165</id>
<summary>[theoraenc] Implement support for new formats</summary>
</bug>
<bug>
<id>594256</id>
<summary>improved slave-skew resynch mechanism</summary>
</bug>
<bug>
<id>594258</id>
<summary>missing break in rtcpbuffer</summary>
</bug>
<bug>
<id>594275</id>
<summary>Add cast to navigation to fix compiler warning</summary>
</bug>
<bug>
<id>594623</id>
<summary>Expose playsink as a fully-fledged element</summary>
</bug>
<bug>
<id>594732</id>
<summary>parse error</summary>
</bug>
<bug>
<id>594757</id>
<summary>build fails due to warning in gstbasertppayload.c</summary>
</bug>
<bug>
<id>594993</id>
<summary>[introspection] pkg-config file madness</summary>
</bug>
<bug>
<id>594994</id>
<summary>[streamvolume] Add get_type function to the documentation</summary>
</bug>
<bug>
<id>595454</id>
<summary>[cddabasesrc] uri format change breaks rhythmbox</summary>
</bug>
<bug>
<id>545807</id>
<summary>[baseaudiosink] audible crack when starting the pipeline</summary>
</bug>
</bugs>
</release>
|
