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authorTim-Philipp Müller <tim@centricular.net>2008-04-07 10:50:11 +0000
committerTim-Philipp Müller <tim@centricular.net>2008-04-07 10:50:11 +0000
commitdf3e7b808a19d0b4ac3ca2c178ebf460998766bc (patch)
tree1a1b4ab7d1b3710b4871781d0243d3e861d4d7ee
parent7a29d716bd8b58d5e1fbcef03e7a0f6c12fbfef3 (diff)
Merge other changes from 0.10.19 release branch.
Original commit message from CVS: * NEWS: * RELEASE: * gst-plugins-base.doap: Merge other changes from 0.10.19 release branch.
-rw-r--r--ChangeLog7
-rw-r--r--NEWS6
-rw-r--r--RELEASE68
-rw-r--r--gst-plugins-base.doap11
4 files changed, 26 insertions, 66 deletions
diff --git a/ChangeLog b/ChangeLog
index 8d5a3261c..564b159cd 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,10 @@
+2008-04-07 Tim-Philipp Müller <tim at centricular dot net>
+
+ * NEWS:
+ * RELEASE:
+ * gst-plugins-base.doap:
+ Merge other changes from 0.10.19 release branch.
+
2008-04-06 Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/audio/gstbaseaudiosink.c:
diff --git a/NEWS b/NEWS
index a973fc59e..17d899d3e 100644
--- a/NEWS
+++ b/NEWS
@@ -1,4 +1,4 @@
-This is GStreamer Base Plug-ins 0.10.18, "I will follow"
+This is GStreamer Base Plug-ins 0.10.19, "Good Times"
IMPORTANT NOTES
@@ -7,6 +7,10 @@ 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.
+Changes since 0.10.18:
+
+ * Handle EAGAIN when polling sockets in rtspconnection
+
Changes since 0.10.17:
* Experimental GIO plugin
diff --git a/RELEASE b/RELEASE
index 00fd777b5..a20820e78 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,5 +1,5 @@
-Release notes for GStreamer Base Plug-ins 0.10.18 "I will follow"
+Release notes for GStreamer Base Plug-ins 0.10.19 "Good Times"
@@ -54,49 +54,9 @@ contains a set of less supported plug-ins that haven't passed the
Features of this release
- * Experimental GIO plugin
- * Continued playbin2 development
- * RTP fixes
- * Better network element support on Windows
- * Various other bug-fixes and improvements
-
-Bugs fixed in this release
-
- * 509637 : [API] [basertpaudiopayload] add _set_samplebits_options()
- * 510229 : [gnomevfssrc] HTTPS support
- * 511478 : [rtpbuffer] add gst_rtp_buffer_set_extension_data function
- * 511810 : [RTSP] Uses MT-unsafe gmtime() function
- * 512899 : [alsa] gstalsasink.c:527: warning: 'snd_pcm_sw_params_set...
- * 513167 : Fix compiler warning due to disabled signals in mixertrac...
- * 514307 : [playbin] warning in nautilus, volume element can't be cr...
- * 514623 : Ogg Theora video slow
- * 514937 : Correct initialization of hints in is_multicast_address()
- * 515654 : xvimagesink doesn't build with --disable-xshm
- * 516246 : [alsasink] handle negative delay from snd_pcm_delay
- * 517420 : typefind: add h264 elementary stream discovery
- * 517991 : problems with configure file depending on GCC compiler
- * 518039 : libgstrtsp MSVC 6.0 compile error
- * 518162 : [subparse] handle italic text starting with " / " with Micr...
- * 518940 : [playbin2] make _get_*_tags() match vfuncs prototype in c...
- * 519906 : [API] add GstMixerOptions::get_values vfunc
- * 519916 : [API] add mixer-changed and options-list-changed messages
- * 520523 : [API] Unreviewed changes to ringbuffer API
- * 521743 : libgstnetbuffer.def exports not up to date
- * 522625 : [video] gst_video_format_parse_caps() broken for RGBA for...
- * 523054 : gstbasesrc crashes when called from typefind helpers
- * 511825 : [RTSP] compiler warning on FreeBSD
- * 520300 : [alsasrc] provide-clock=false messes up buffer durations
-
-API changed in this release
-
-- API additions:
+ * Handle EAGAIN when polling sockets in rtspconnection
+There were no bugs fixed in this release
-* GstRTPBuffer:gst_rtp_buffer_set_extension_data()
-* add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
-* add GstMixerOptions::get_values vfunc (#519906)
-* add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
-* gst_base_rtp_audio_payload_set_samplebits_options()
-* GstNetBuffer::gst_netaddress_equal
Download
@@ -126,28 +86,6 @@ Applications
Contributors to this release
- * Alan Peevers
- * Branko Čibej
- * Brian Cameron
- * David Schleef
* Jan Schmidt
- * Josep Torra Valles
- * José Alburquerque
- * Julien Moutte
- * Mersad Jelacic
- * Michael Smith
- * Milosz Derezynski
- * Ole André Vadla Ravnås
- * Olivier Crete
- * Peter Kjellerstedt
- * Rene Stadler
- * Sebastian Dröge
- * Stefan Kost
- * Thijs Vermeir
- * Tim-Philipp Müller
- * Tomasz Sałaciński
- * Tommi Myöhänen
- * William M. Brack
* Wim Taymans
- * Zaheer Abbas Merali
  \ No newline at end of file
diff --git a/gst-plugins-base.doap b/gst-plugins-base.doap
index cea5e9a9f..22f3f752b 100644
--- a/gst-plugins-base.doap
+++ b/gst-plugins-base.doap
@@ -37,6 +37,17 @@ A wide range of video and audio decoders, encoders, and filters are included.
<release>
<Version>
+ <revision>0.10.19</revision>
+ <branch>0.10</branch>
+ <name>Good Times</name>
+ <created>2008-04-04</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-0.10.19.tar.bz2" />
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-0.10.19.tar.gz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>0.10.18</revision>
<branch>0.10</branch>
<name>I will follow</name>