From 225d6d77b172eb982eebdd092e378c21896208ea Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Mon, 5 Oct 2009 13:56:15 +0100 Subject: Release 0.10.25 --- NEWS | 80 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 79 insertions(+), 1 deletion(-) (limited to 'NEWS') diff --git a/NEWS b/NEWS index 93b37c6e9..d6eec5e08 100644 --- a/NEWS +++ b/NEWS @@ -1,4 +1,82 @@ -This is GStreamer Base Plug-ins 0.10.24, "Counting the days" +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 Changes since 0.10.23: -- cgit v1.2.3