Commit b4772b4c authored by Tim-Philipp Müller's avatar Tim-Philipp Müller
Browse files

Release 1.0.2

parent 45d802b6
=== release 1.0.2 ===
2012-10-25 Tim-Philipp Müller <tim@centricular.net>
* configure.ac:
releasing 1.0.2
2012-10-24 14:05:56 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: track forced decoding state
2012-10-24 13:34:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Also send a GAP event to let audio sinks start their clock in case they did not have enough data yet
2012-10-24 13:29:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Use correct timestamp/duration for the GAP events
2012-10-24 13:26:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* ext/alsa/Makefile.am:
* ext/cdparanoia/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* ext/pango/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst/adder/Makefile.am:
* gst/app/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/audiotestsrc/Makefile.am:
* gst/encoding/Makefile.am:
* gst/gio/Makefile.am:
* gst/playback/Makefile.am:
* gst/subparse/Makefile.am:
* gst/tcp/Makefile.am:
* gst/typefind/Makefile.am:
* gst/videoconvert/Makefile.am:
* gst/videorate/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/videotestsrc/Makefile.am:
* gst/volume/Makefile.am:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
Revert "gst: Add better support for static plugins"
This reverts commit d2d79e3bc2a02ec57258e504b031f7e2d3729ea2,
which was accidentially pushed.
2012-10-24 13:25:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Send GAP events to advance streams
2012-10-24 12:10:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* ext/alsa/Makefile.am:
* ext/cdparanoia/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* ext/pango/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst/adder/Makefile.am:
* gst/app/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/audiotestsrc/Makefile.am:
* gst/encoding/Makefile.am:
* gst/gio/Makefile.am:
* gst/playback/Makefile.am:
* gst/subparse/Makefile.am:
* gst/tcp/Makefile.am:
* gst/typefind/Makefile.am:
* gst/videoconvert/Makefile.am:
* gst/videorate/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/videotestsrc/Makefile.am:
* gst/volume/Makefile.am:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
gst: Add better support for static plugins
2012-10-24 11:22:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudiobasesink.c:
audiobasesink: Add explanation to the GAP event handling code
2012-10-24 09:57:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Create a GAP event with a sensible timestamp
2012-10-24 11:16:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudiobasesink.c:
audiobasesink: Properly handle GAP events
These are now converted into silence buffers if they have
a duration or cause the ringbuffer and clock to be started
if they don't have a duration.
Fixes bug #685273.
2012-10-23 18:16:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Also propagate return value of pushing GAP event upstream
2012-10-23 17:37:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Return TRUE from the EOS handler
2012-10-23 15:56:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/gstvorbistag.c:
vorbistag: add mapping for 'ALBUM ARTIST' with space
As found in sample file for bug #684701.
2012-10-22 15:44:16 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/tcp/gstmultihandlesink.c:
tcp: sys/socket.h is needed for getsockname() and similar functions
2012-10-22 10:30:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/riff/riff-media.c:
riff: add bpp to caps for msvideo
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686298
2012-10-22 09:44:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/videoconvert/videoconvert.c:
videoconvert: add more debug
2012-10-20 12:59:11 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst-libs/gst/tag/mklicensestables.c:
tag: remove unnecessary g_type_init() call from mklicensestable tool
https://bugzilla.gnome.org/show_bug.cgi?id=686456
2012-10-20 11:38:55 +0100 Tim-Philipp Müller <tim@centricular.net>
* ext/alsa/gstalsasink.c:
alsasink: fix caps leak in acceptcaps function
https://bugzilla.gnome.org/show_bug.cgi?id=681192
2012-10-20 11:38:10 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: don't leak message strings when error is not fatal
https://bugzilla.gnome.org/show_bug.cgi?id=681192
2012-10-20 11:37:33 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst-libs/gst/video/gstvideodecoder.c:
videodecoder: don't leak message strings when error is not fatal
2012-10-19 18:29:00 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
tcpserver{sink,src}: improve docs and property strings
And some minor clean-ups.
2012-10-17 12:19:56 +0200 Alexandre Relange <alexandre.relange@pineasystems.org>
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.c:
* gst/tcp/gsttcpserversrc.h:
tcpserver{sink,src}: add 'current-port' property and signal actually used port
Useful when port=0 (use random available port) was requested.
https://bugzilla.gnome.org/show_bug.cgi?id=580093
2012-10-18 22:13:09 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/audioconvert/gstaudioconvert.c:
audioconvert: enhance transforming caps
... so as to preserve input format precision,
and preferably not convert at all.
2012-10-18 12:02:00 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
* gst-libs/gst/tag/gstvorbistag.c:
vorbistag: fix 'TODO' on image tag parsing
Image tag now uses GstSample that has the buffer and caps
associated with it.
2012-10-18 00:39:42 +0100 Tim-Philipp Müller <tim@centricular.net>
* ext/alsa/gstalsa.c:
alsa: if no formats in native endianness could be detected, try non-native endianness as well
This can happen, e.g. when using an USB sound card on
a big-endian device
https://bugzilla.gnome.org/show_bug.cgi?id=680904
2012-10-18 00:04:06 +0100 Tim-Philipp Müller <tim@centricular.net>
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsasink.c:
alsa: fix supported format detection
The format probing code was assuming there'd be one caps
structure for each separate width/depth combination like
we did in 0.10 all over the place: for one, we'd query
unsigned/signed formats together for the same width/height,
and we'd add the entire current structure to the probed
caps when we find a format is supported. Now that we have
all raw formats in a single structure, this is all not going
to work so well any more. We added the entire structure with
all possible formats to the caps if we support just one format.
Fix probing so that we only return the list of actually
supported raw audio formats (with native endianness) from
get_caps().
2012-10-17 19:59:57 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst-libs/gst/audio/gstaudiocdsrc.c:
* gst-libs/gst/audio/gstaudiocdsrc.h:
audiocdsrc: mention TOCs in docs
2012-10-17 16:54:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoraenc.c:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
theora, app: use gst_element_class_set_static_metadata()
Avoids string copies.
2012-10-17 10:55:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/video/gstvideodecoder.c:
videodecoder: return NULL from _allocate_output_buffer() if alloc fails
.. instead of garbage pointer. Also log failure in debug log.
Should've returned the flow return like _allocate_output_frame().
https://bugzilla.gnome.org/show_bug.cgi?id=683098
2012-10-16 11:48:32 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst-libs/gst/riff/riff-media.c:
riff-media: fix palette extraction some more
We still need to make sure the palette is always at least 1024
bytes.
2012-10-16 00:55:56 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst-libs/gst/riff/riff-media.c:
riff: create palette_data buffer correctly
gst_buffer_copy_into() will append to any existing
memory region, so don't create a buffer and alloc
some memory, but just create an empty buffer and
let _copy_into() append the memory we want. Fixes
the palette being 2048 bytes with the first half
being filled with garbage.
https://bugzilla.gnome.org/show_bug.cgi?id=686046
2012-10-15 18:47:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/audio/audio.c:
audio: properly handle clipping of empty buffer
2012-10-15 16:33:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
videotestsrc: make and copy palette
2012-10-15 16:32:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/videoconvert/videoconvert.c:
videoconvert: actually copy the palette
Copy the default palette in the destination buffer too.
2012-10-15 15:50:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/design/part-mediatype-video-raw.txt:
docs: fix RGB8P format description docs
2012-10-11 11:36:54 +0200 David Corvoysier <david.corvoysier@orange.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Fix group switching algorithm
There were two issues with the previous decodebin2 group switching algorithm:
Issue 1: It operated with no memory of what has been drained or not, leading to
multiple checks for chains/groups that were already drained.
Issue 2: When receiving an EOS, it only detected that a higher-level chain
was drained if it contained the pad receiving the EOS.
The following modifications have been applied:
- a new drained property has been added to GstDecodeChain
- both drained properties of chain/group are set as soon as they are detected
- the algorithm now tests agains these values
See https://bugzilla.gnome.org/show_bug.cgi?id=685938
2012-09-20 01:07:08 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst-libs/gst/rtsp/gstrtsprange.c:
rtsprange: fix formatting and parsing of range floating-point values
Other locales might use a comma instead of a floating point
for floats, which might lead to parsing errors.
https://bugzilla.gnome.org/show_bug.cgi?id=684411
2012-10-12 21:36:49 +0100 Tim-Philipp Müller <tim@centricular.net>
* docs/design/part-mediatype-video-raw.txt:
docs: update for RGB8_PALETTED -> RGB8P
2012-10-12 21:31:25 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst-libs/gst/riff/riff-media.c:
riff: 8-bit paletted video is format RGB8P, not RGB8_PALETTED
https://bugzilla.gnome.org/show_bug.cgi?id=686046
2012-10-11 12:54:39 +0200 Josep Torra <n770galaxy@gmail.com>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: set of base_ts for segment formats other than time
Fixes setting of converted segment start as base_ts when estimate rate
is allowed.
2012-10-10 15:49:46 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: Don't unref caps twice
Thanks to Josep Torra for noticing.
2012-10-10 15:04:07 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst-libs/gst/video/gstvideodecoder.c:
* gst-libs/gst/video/gstvideoutils.h:
videodecoder: finetune missing timestamp estimating
Monitor for reordered output timestamps, and then avoid oldest DTS
as PTS approach, and try for an oldest PTS as out PTS approach,
if at least all valid PTS available.
Avoids bogus estimating upon sparse available input PTS, and tries
to handle all-keyframe input, or input PTS which are actually DTS.
2012-10-10 11:50:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/playback/gstplaysinkconvertbin.c:
playsinkconvertbin: Change GST_WARNING to GST_INFO
It's not a problem if we have no converters, this only means
that none were requested at this point.
2012-10-09 13:07:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/vorbis/gstivorbisdec.c:
* ext/vorbis/gstvorbisdec.c:
ivorbisdec: Rename debug category to prevent symbol conflict when using static linking
2012-10-09 12:18:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
* gst-libs/gst/audio/streamvolume.c:
* gst/playback/gstplaybin2.c:
* tests/examples/app/appsrc-ra.c:
* tests/examples/app/appsrc-seekable.c:
* tests/examples/app/appsrc-stream.c:
* tests/examples/app/appsrc-stream2.c:
* tests/examples/gio/giosrc-mounting.c:
docs: playbin2 -> playbin
2012-10-09 12:17:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/app/appsink-src.c:
tests: fix audio caps
2012-10-08 12:43:03 +0200 Andoni Morales Alastruey <ylatuya@gmail.com>
* gst-libs/gst/audio/gstaudiodecoder.h:
* gst-libs/gst/audio/gstaudioencoder.h:
* gst-libs/gst/video/gstvideodecoder.h:
* gst-libs/gst/video/gstvideoencoder.h:
audio/video: update documentation for vfunc's that require chaining up
2012-10-07 02:58:05 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
* configure.ac:
configure: Reintroduced xmmintrin.h/emmintrin.h header checks
The audio resampler needs these for the SSE/SSE2 code paths
Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
2012-10-08 09:21:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/video/gstvideodecoder.h:
video: small docs fix
2012-10-07 19:46:45 +0100 Tim-Philipp Müller <tim@centricular.net>
* tests/check/libs/video.c:
tests: fix video overlay_composition_premultiplied_alpha test on big-endian machines
The unit test was checking for alpha at the wrong position.
2012-10-07 16:52:27 +0100 Tim-Philipp Müller <tim@centricular.net>
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-ivorbisdec.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/_stdint.h:
* win32/common/config.h:
Back to development (bug fixing)
=== release 1.0.1 ===
 
2012-10-07 Tim-Philipp Müller <tim@centricular.net>
2012-10-07 15:11:10 +0100 Tim-Philipp Müller <tim@centricular.net>
 
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
releasing 1.0.1
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-ivorbisdec.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 1.0.1
 
2012-10-07 13:34:06 +0100 Tim-Philipp Müller <tim@centricular.net>
 
This is GStreamer Base Plugins 1.0.1
This is GStreamer Base Plugins 1.0.2
Changes since 1.0.1:
* Parallel installability with 0.10.x series
* alsa: fix probing of supported formats, and advertise non-native-endianness formats as fallback
* audiobasesink: properly handle GAP events (fixing some isses with e.g. certain DVD menus)
* audioconvert: try harder to not convert or to preserve input format precision
* audiodecoder: leak fixes and refcounting fixes
* audioresample: re-enable the SSE/SSE2 code paths for better performance
* riff: fix paletted RGB formats and msvideo mapping
* rtsp: make formatting and parsing of range floating-point values locale-independent
* playbin: streamsynchronizer fixes, esp. for handling corner-cases near EOS
* tcpserver{sink,src}: add 'current-port' property and signal actually used port
* videoconvert: fix handling of paletted RGB formats
* videodecoder: don't leak message strings when error is not fatal
* videodecoder: finetune missing timestamp estimating
* videotestsrc: add palette for paletted RGB formats
* vorbistag: fix writing of image tags into vorbis comments
Bugs fixed since 1.0.1:
* 580093 : tcpserversink,src: add 'current-port' property and signal actually used port when port=0 was set
* 680904 : alsasink: no supported formats detected with using USB sound card on big-endian system
* 683098 : videodecoder: log failure message if acquire_buffer failed
* 684411 : rtsp: range in SDP formatted according to locale
* 685273 : Pre-rolling on GAP events doesn't work properly for audio sinks
* 685711 : audio, video: update docs for virtual functions that require chaining up
* 685938 : [decodebin] Issues with group switching algorithm
* 686081 : adder: all unit tests crash now after collectpads changes
* 686298 : Cannot decode some AVI files with Microsoft Video 1
Changes since 1.0.0:
......
Release notes for GStreamer Base Plugins 1.0.1
Release notes for GStreamer Base Plugins 1.0.2
The GStreamer team is proud to announce a new release
The GStreamer team is proud to announce a new bug-fix release
in the 1.x stable series of the
core of the GStreamer streaming media framework.
......@@ -61,17 +61,32 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
Features of this release
* Parallel installability with 0.10.x series
* videodecoder and -encoder timestamp handling improvements
* thread-safey fixes for GstMeta registrations and GstVideoDecoder
* alsa: fix probing of supported formats, and advertise non-native-endianness formats as fallback
* audiobasesink: properly handle GAP events (fixing some isses with e.g. certain DVD menus)
* audioconvert: try harder to not convert or to preserve input format precision
* audiodecoder: leak fixes and refcounting fixes
* audioresample: re-enable the SSE/SSE2 code paths for better performance
* riff: fix paletted RGB formats and msvideo mapping
* rtsp: make formatting and parsing of range floating-point values locale-independent
* playbin: streamsynchronizer fixes, esp. for handling corner-cases near EOS
* tcpserver{sink,src}: add 'current-port' property and signal actually used port
* videoconvert: fix handling of paletted RGB formats
* videodecoder: don't leak message strings when error is not fatal
* videodecoder: finetune missing timestamp estimating
* videotestsrc: add palette for paletted RGB formats
* vorbistag: fix writing of image tags into vorbis comments
Bugs fixed in this release
* 684424 : playbin: external subtitles break playback
* 684832 : videodecoder: Takes stream lock in query function
* 685110 : encodebin fails to release mux request sink pad for GstId3Mux as it is a static one for this mux
* 685242 : rtsp: mark url argument of gst_rtsp_url_parse as out
* 685332 : GstMeta registry race
* 685490 : audioencoder: don't require base class to implement to start vfunc
* 580093 : tcpserversink,src: add 'current-port' property and signal actually used port when port=0 was set
* 680904 : alsasink: no supported formats detected with using USB sound card on big-endian system
* 683098 : videodecoder: log failure message if acquire_buffer failed
* 684411 : rtsp: range in SDP formatted according to locale
* 685273 : Pre-rolling on GAP events doesn't work properly for audio sinks
* 685711 : audio, video: update docs for virtual functions that require chaining up
* 685938 : [decodebin] Issues with group switching algorithm
* 686081 : adder: all unit tests crash now after collectpads changes
* 686298 : Cannot decode some AVI files with Microsoft Video 1
==== Download ====
......@@ -108,13 +123,14 @@ subscribe to the gstreamer-devel list.
Contributors to this release
* Alban Browaeys
* Alexandre Relange
* Andoni Morales Alastruey
* Carlos Rafael Giani
* David Corvoysier
* Josep Torra
* Mark Nauwelaerts
* Michael Smith
* Olivier Crête
* Sebastian Dröge
* Sebastian Pölsterl
* Thiago Santos
* Tim-Philipp Müller
* Wim Taymans
 
\ No newline at end of file
......@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
dnl initialize autoconf
dnl releases only do -Wall, git and prerelease does -Werror too
dnl use a three digit version number for releases, and four for git/prerelease
AC_INIT(GStreamer Base Plug-ins, 1.0.1.1,
AC_INIT(GStreamer Base Plug-ins, 1.0.2,
http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer,
gst-plugins-base)
......@@ -50,7 +50,7 @@ dnl - interfaces added/removed/changed -> increment CURRENT, REVISION = 0
dnl - interfaces added -> increment AGE
dnl - interfaces removed -> AGE = 0
dnl sets GST_LT_LDFLAGS
AS_LIBTOOL(GST, 1, 0, 1)
AS_LIBTOOL(GST, 2, 0, 2)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.0.0
......
......@@ -664,7 +664,7 @@
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>host</NICK>
<BLURB>The host/IP to send the packets to.</BLURB>
<BLURB>The host/IP to listen on.</BLURB>
<DEFAULT>"localhost"</DEFAULT>
</ARG>
......@@ -674,10 +674,20 @@
<RANGE>[0,65535]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>port</NICK>
<BLURB>The port to send the packets to.</BLURB>
<BLURB>The port to listen to (0=random available port).</BLURB>
<DEFAULT>4953</DEFAULT>
</ARG>
<ARG>
<NAME>GstTCPServerSink::current-port</NAME>
<TYPE>gint</TYPE>
<RANGE>[0,65535]</RANGE>
<FLAGS>r</FLAGS>
<NICK>current-port</NICK>
<BLURB>The port number the socket is currently bound to.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstTCPServerSrc::host</NAME>
<TYPE>gchar*</TYPE>
......@@ -694,7 +704,7 @@
<RANGE>[0,65535]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Port</NICK>
<BLURB>The port to listen to.</BLURB>
<BLURB>The port to listen to (0=random available port).</BLURB>
<DEFAULT>4953</DEFAULT>
</ARG>
......@@ -708,6 +718,16 @@
<DEFAULT>GST_TCP_PROTOCOL_NONE</DEFAULT>
</ARG>
<ARG>
<NAME>GstTCPServerSrc::current-port</NAME>
<TYPE>gint</TYPE>
<RANGE>[0,65535]</RANGE>
<FLAGS>r</FLAGS>
<NICK>current-port</NICK>
<BLURB>The port number the socket is currently bound to.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstMultiFdSink::buffers-max</NAME>
<TYPE>gint</TYPE>
......
......@@ -3,10 +3,10 @@
<description>Adds multiple streams</description>
<filename>../../gst/adder/.libs/libgstadder.so</filename>
<basename>libgstadder.so</basename>
<version>1.0.1.1</version>