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Release 1.1.2
This commit is contained in:
Sebastian Dröge
parent
a9b4801cfb
commit
96f686fc1f
34 changed files with 563 additions and 175 deletions
494
ChangeLog
494
ChangeLog
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@ -1,9 +1,497 @@
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=== release 1.1.1 ===
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=== release 1.1.2 ===
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2013-06-05 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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2013-07-11 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* configure.ac:
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releasing 1.1.1
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releasing 1.1.2
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2013-07-10 17:16:14 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gstplaybin2.c:
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playbin: Only give sinks a new bus if they have no parent yet
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Otherwise we will remove the bus that would proxy messages to playsink
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and never set it again. If the sink is already in playsink, all failures
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are fatal anyway as it's either a sink that worked before or one that
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was set by the user.
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https://bugzilla.gnome.org/show_bug.cgi?id=701997
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2013-07-10 13:22:04 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gstplaybin2.c:
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playbin: Store a/v/t sinks locally too, not just in playsink
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2013-07-10 13:21:29 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gstplaysink.c:
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playsink: ref_sink() any sinks that are set on playsink
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Otherwise the behaviour of the properties is inconsistent.
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2013-07-10 13:20:34 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* tests/check/elements/playbin.c:
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playbin: Fix assumptions in the unit test
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Unused sinks are still set to READY now during autoplugging
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to check their caps. Also playsink owns a ref to the sinks too.
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2013-07-10 13:00:21 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gststreamsynchronizer.c:
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streamsynchronizer: Non-TIME segment streams are not waiting automatically
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This was leftover code from porting to 1.0 and fixes the playbin
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unit test.
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https://bugzilla.gnome.org/show_bug.cgi?id=701943
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2013-07-09 23:04:49 +0200 Branko Subasic <branko@axis.com>
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* win32/common/libgstrtp.def:
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win32: add missing rtp buffer methods
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2013-07-09 14:55:57 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gstplaybin2.c:
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* gst/playback/gstplaysink.c:
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playbin: Change sink ownership handling to be a bit more sane
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playbin will now only activate the sinks in a single place and
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will never change the states of any sinks that are owned by
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playsink.
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Also handle text-sinks the same way as audio/video sinks inside
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playbin.
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2013-07-05 21:55:26 +0200 Piotr Drąg <piotrdrag@gmail.com>
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* po/POTFILES.in:
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po: update POTFILES.in
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https://bugzilla.gnome.org/show_bug.cgi?id=703684
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2013-07-04 17:09:00 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
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* gst-libs/gst/video/colorbalance.c:
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colorbalance: Fix the typo in base_init().
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2013-07-04 12:54:59 -0400 Thibault Saunier <thibault.saunier@collabora.com>
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* gst/adder/gstadder.c:
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adder: Do not send flush_start event with the stream lock taken
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FLUSH_START is not serialized, so the lock should not be taken when
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sending it.
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2013-07-05 00:47:08 +0100 Marcin Lewandowski <marcin@saepia.net>
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* gst-libs/gst/tag/id3v2frames.c:
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tag: ignore malformed ID3v2 TDAT frames
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Just skip them, don't cause criticals.
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https://bugzilla.gnome.org/show_bug.cgi?id=703283
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2013-07-03 09:44:32 +0100 Tim-Philipp Müller <tim@centricular.net>
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* gst/audioresample/speex_resampler_int.c:
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audioresample: make explicit that neon is disabled and why
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https://bugzilla.gnome.org/show_bug.cgi?id=703477
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2013-07-02 18:20:39 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
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* gst/audioresample/speex_resampler_int.c:
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audioresample: disable 16-bit integer NEON support
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it seems to be broken (produces no audio), plus the performance gain
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is small
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Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
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2013-07-02 14:25:28 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gstplaybin2.c:
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playbin: If we had a previous autoplugged sink, try to reuse it
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https://bugzilla.gnome.org/show_bug.cgi?id=701997
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2013-07-02 14:18:20 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gstplaysink.c:
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playsink: If we switch sinks, make sure that the old sink is set to NULL
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2013-07-02 14:02:57 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gstplaybin2.c:
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playbin: Don't change the state of sinks that we passed to playsink already
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2013-07-02 14:01:52 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gstplaysink.c:
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playsink: Consider new audio/video sinks when reconfiguring
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2013-07-02 12:27:03 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gstplaybin2.c:
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playbin: Improve debug output regarding sink selection
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2013-07-01 12:52:43 -0600 Brendan Long <self@brendanlong.com>
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* gst/playback/gstplaybin2.c:
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playbin: Post an error message if a stream combiner doesn't return a request pad.
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2013-07-01 13:45:25 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gstplaybin2.c:
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playbin: Only intersect to check if a sink can handle raw caps
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Doing a subset check requires fixed caps, which we might not have here.
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https://bugs.webkit.org/show_bug.cgi?id=116042
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2013-07-01 10:39:02 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
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* gst-libs/gst/pbutils/descriptions.c:
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* gst-libs/gst/pbutils/missing-plugins.c:
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* gst-libs/gst/pbutils/pbutils-private.h:
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pbutils: allow describing unfixed caps if they share the same media type
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Caps description and missing plugin code does not really need caps to
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be fixed, and indeed they may not be if giving encodebin unfixed caps
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that correspond to an unknown encoder or muxer.
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So we relax the check, and allow unfixed caps if all the structures
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refer to the same media type.
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2013-07-01 11:16:34 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst-libs/gst/video/gstvideodecoder.c:
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videodecoder: Send all pending events with type < CAPS before sending caps
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2013-06-27 16:33:15 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
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* gst-libs/gst/video/gstvideoencoder.c:
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videoencoder: Send all pending events with type < CAPS before sending caps.
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https://bugzilla.gnome.org/show_bug.cgi?id=703196
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2013-06-28 14:48:19 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
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* gst/typefind/gsttypefindfunctions.c:
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typefind: avoid too low mpeg/ts probability on small amount of data
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With the current test, we get into problems when we try to typefind
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a MPEG stream from a small amount of data, which can happen when
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we get data pushed from a HTTP source. We thus make a second test
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to give higher probability if all the potential headers were either
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pack or pes headers (ie, no potential header was unrecognized).
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This fixes an issue with a MPEG1/MP2 stream being properly discovered
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as video/mpeg from a file, but as audio/mpeg from souphttpsrc.
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https://bugzilla.gnome.org/show_bug.cgi?id=703256
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2013-06-30 18:17:15 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst-libs/gst/video/gstvideodecoder.c:
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* gst-libs/gst/video/gstvideoencoder.c:
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video(enc|dec)oder: Don't return not-negotiated if flushing
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If the pad is flushing after a failed negotiation, return
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GST_FLOW_FLUSHING instead from finish_frame().
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https://bugzilla.gnome.org/show_bug.cgi?id=701763
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2013-06-30 18:16:35 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst-libs/gst/audio/gstaudioencoder.c:
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audioencoder: Don't return not-negotiated if flushing
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If the pad is flushing after a failed negotiation, return
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GST_FLOW_FLUSHING instead from finish_frame().
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https://bugzilla.gnome.org/show_bug.cgi?id=701763
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2013-06-14 07:23:40 +0200 Edward Hervey <edward@collabora.com>
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* gst-libs/gst/pbutils/descriptions.c:
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* tests/check/libs/pbutils.c:
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pbutils: descriptions: Allow smart codec tag handling
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We already have internally the information on what type of stream (audio,
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video, container, subtitle, ...) a certain caps is.
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Instead of forcing callers to specify which CODEC_TAG category a certain
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caps is, use that information to make a smart choice.
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Does not break previous behaviour of gst_pb_utils_add_codec_description_to_tag_list
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(if tag is specified it will be used, if caps is invalid it will be rejected,
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...).
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https://bugzilla.gnome.org/show_bug.cgi?id=702215
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2013-06-19 09:25:48 +0200 Edward Hervey <edward@collabora.com>
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* gst-libs/gst/tag/gstxmptag.c:
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xmptag: Add a debug category
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Instead of using the default category
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2013-06-27 12:23:27 +0200 Patricia Muscalu <patricia@axis.com>
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* gst/videotestsrc/gstvideotestsrc.c:
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videotestsrc: do not leak lines
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703177
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2013-06-26 14:36:17 +0200 Ognyan Tonchev <ognyan@axis.com>
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* gst-libs/gst/rtp/gstrtpbasepayload.c:
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rtpbasepayload: Do not leak the event when segment is delayed
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703119
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2013-06-26 15:03:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst-libs/gst/rtsp/gstrtspconnection.c:
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rtsp: make read uncancelable when reading a message
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When we start to read a message, we need to continue reading until the end of
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the message or else we lose track and cause parse errors. Use a variable
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may_cancel to avoid cancelation after we read the first byte until we have
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the complete message.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703088
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2013-06-21 20:41:15 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
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* gst-libs/gst/audio/gstaudiodecoder.c:
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audiodecoder: Don't return not-negotiated if flushing
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If the pad is flushing after a failed negotiation, return GST_FLOW_FLUSHING.
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https://bugzilla.gnome.org/show_bug.cgi?id=701763
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2013-06-23 12:07:41 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* ext/ogg/gstoggstream.c:
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ogg: The Daala headers are little endian, not big endian
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2013-06-23 10:30:02 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* ext/ogg/gstoggmux.c:
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* ext/ogg/gstoggstream.c:
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ogg: Add Daala support
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2013-06-21 19:04:43 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst-libs/gst/pbutils/descriptions.c:
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pbutils: Add VP9 description
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2013-06-17 08:58:13 +0200 Edward Hervey <edward@collabora.com>
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* gst-libs/gst/video/gstvideodecoder.c:
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videodecoder: Fix drop frame handling at startup
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In the unlikely case that the decoder drops a frame before the first
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input frame is outputted, use the input segment (since it wasn't
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carried over to the output segment yet)
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https://bugzilla.gnome.org/show_bug.cgi?id=702502
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2013-06-21 11:50:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst-libs/gst/rtsp/gstrtspconnection.c:
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rtsp: dispatch when initial buffer has data
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When we have data in the inital buffer, dispath the read function to read it
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even if the socket has no data to read.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702652
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2013-06-20 17:28:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst-libs/gst/rtsp/gstrtspconnection.c:
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rtsp: manage writer child source better
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Only add the write child source when we have something to write or else
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we will dispatch forever without doing anything.
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2013-06-19 13:21:45 +0200 Jonas Holmberg <jonashg@axis.com>
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* gst-libs/gst/audio/gstaudioencoder.c:
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audioencoder: unref before memset
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Unref allocator and input_caps in encoder context before memsetting the
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context.
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2013-06-19 09:22:50 +0200 Edward Hervey <edward@collabora.com>
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* gst-libs/gst/tag/gstxmptag.c:
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xmptag: More efficient GSList usage
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Instead of constantly appending (which gets more and more expensive), just
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prepend to the list (O(1)) and reverse the list before usage.
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https://bugzilla.gnome.org/show_bug.cgi?id=702545
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2013-06-16 22:39:30 +0200 Branko Subasic <branko@axis.com>
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* gst-libs/gst/rtp/gstrtpbuffer.c:
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* gst-libs/gst/rtp/gstrtpbuffer.h:
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* tests/check/libs/rtp.c:
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rtpbuffer: add gst_rtp_buffer_get_payload_bytes
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The function gst_rtp_buffer_get_payload can not be used in Python
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because it lacks necessary length parameter. This patch adds a new
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function, gst_rtp_buffer_get_payload_bytes, to use from Python
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bindings. The new function has the advisory "Rename to:" annotation
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so it can replace the gst_rtp_buffer_get_payload whan creating
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bindings.
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The function gst_rtp_buffer_get_extension_bytes is also added. It wraps
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gst_rtp_buffer_get_extension_data which doesn't work in Python due to
|
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incomplete annotation and because it returns the length as number of
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32-bit words.
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https://bugzilla.gnome.org/show_bug.cgi?id=698562
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2013-06-17 16:34:26 +0200 Ognyan Tonchev <ognyan@axis.com>
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* gst-libs/gst/audio/gstaudiobasesrc.c:
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audiobasesrc: add 2 missing gst_buffer_unmap () calls
|
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There are 2 missing calls to gst_buffer_unmap () in the error handling in
|
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create ().
|
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702467
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2013-06-17 16:02:41 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
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* gst/playback/gstplaysink.c:
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playsink: Fix the block diagram of deinterlace bin.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702465
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2013-06-13 11:08:20 -0600 Brendan Long <b.long@cablelabs.com>
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|
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* gst/playback/gstplaybin2.c:
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playbin: Emit {audio,text,video}-changed signals when pads are removed
|
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https://bugzilla.gnome.org/show_bug.cgi?id=702195
|
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|
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2013-06-11 15:22:50 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/videoconvert/videoconvert.c:
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videoconvert: Fix leaking of the chroma resample helper objects
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|
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2013-06-10 14:43:35 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
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* tests/check/Makefile.am:
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* tests/check/elements/playbin-complex.c:
|
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tests: add more unit test for playbin
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Add unit test for autoplugging of video_decoder/video_sink combination
|
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based on capsfeatures.
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2013-06-10 15:31:38 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst-libs/gst/rtsp/gstrtspconnection.c:
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rtspconnection: Make sure to set a sensible default port for the GSocketConnection
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Otherwise it will connect to port 0 if no port is given in the URI.
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https://bugzilla.gnome.org/show_bug.cgi?id=701798
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|
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2013-06-09 19:20:20 +0200 Sebastian Dröge <slomo@circular-chaos.org>
|
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|
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* gst/adder/gstadder.c:
|
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adder: Reject segments that have a different rate than the output segment
|
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adder does no rate conversion.
|
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|
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2013-06-08 23:51:13 +0200 Sebastian Dröge <slomo@circular-chaos.org>
|
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|
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* gst/playback/gstplaybin2.c:
|
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playbin: When activating a fixed sink, proxy error messages too
|
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If activating a fixed sink fails, everything will fail later anyway
|
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and we can just error out early.
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2013-06-08 23:34:53 +0200 Sebastian Dröge <slomo@circular-chaos.org>
|
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|
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* gst/playback/gstplaybin2.c:
|
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playbin: Improve autoplugging of decoder/sink combinations by trying to activate the sink
|
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And if that fails don't bother autoplugging that sink. Also gives
|
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us more accurate sink caps.
|
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2013-06-08 23:08:05 +0200 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/playback/gstplaybin2.c:
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playbin: Proxy the playbin context to the sinks
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2013-06-08 23:04:43 +0200 Sebastian Dröge <slomo@circular-chaos.org>
|
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|
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* gst/playback/gstplaybin2.c:
|
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playbin: Proxy sink messages if we activate a sink in playbin already
|
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This makes sure the application gets any context related messages and
|
||||
can do whatever is required to a) get the sink a context or b) share
|
||||
the context with other elements in the pipeline.
|
||||
The proxying is necessary because the sink is not a child element of
|
||||
playbin, but instead will at a later point be a child of some bin
|
||||
inside playsink.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=700967
|
||||
|
||||
2013-06-06 15:57:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
||||
|
||||
* gst/playback/gstdecodebin2.c:
|
||||
decodebin: Let serialize queries before caps events through
|
||||
Otherwise we're going to deadlock forever because no autoplugging
|
||||
happens without having caps, but caps can never be send because
|
||||
we're blocking.
|
||||
Serialized queries before caps should never be sent unless really
|
||||
necessary.
|
||||
|
||||
2013-06-05 18:36:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
||||
|
||||
* configure.ac:
|
||||
Back to development
|
||||
|
||||
=== release 1.1.1 ===
|
||||
|
||||
2013-06-05 17:58:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
||||
|
||||
* ChangeLog:
|
||||
* NEWS:
|
||||
* RELEASE:
|
||||
* common:
|
||||
* configure.ac:
|
||||
* docs/plugins/gst-plugins-base-plugins.args:
|
||||
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
||||
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
||||
* docs/plugins/gst-plugins-base-plugins.signals:
|
||||
* 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-libs/gst/audio/gstaudiopack-dist.c:
|
||||
* gst-libs/gst/video/video-orc-dist.c:
|
||||
* gst-libs/gst/video/video-orc-dist.h:
|
||||
* gst-plugins-base.doap:
|
||||
* gst/audioconvert/gstaudioconvertorc-dist.c:
|
||||
* gst/videoconvert/gstvideoconvertorc-dist.c:
|
||||
* gst/videoscale/gstvideoscaleorc-dist.c:
|
||||
* gst/volume/gstvolumeorc-dist.c:
|
||||
* po/af.po:
|
||||
* po/az.po:
|
||||
* po/bg.po:
|
||||
* po/ca.po:
|
||||
* po/cs.po:
|
||||
* po/da.po:
|
||||
* po/de.po:
|
||||
* po/el.po:
|
||||
* po/en_GB.po:
|
||||
* po/eo.po:
|
||||
* po/es.po:
|
||||
* po/eu.po:
|
||||
* po/fi.po:
|
||||
* po/fr.po:
|
||||
* po/gl.po:
|
||||
* po/hu.po:
|
||||
* po/id.po:
|
||||
* po/it.po:
|
||||
* po/ja.po:
|
||||
* po/lt.po:
|
||||
* po/lv.po:
|
||||
* po/nb.po:
|
||||
* po/nl.po:
|
||||
* po/or.po:
|
||||
* po/pl.po:
|
||||
* po/pt_BR.po:
|
||||
* po/ro.po:
|
||||
* po/ru.po:
|
||||
* po/sk.po:
|
||||
* po/sl.po:
|
||||
* po/sq.po:
|
||||
* po/sr.po:
|
||||
* po/sv.po:
|
||||
* po/tr.po:
|
||||
* po/uk.po:
|
||||
* po/vi.po:
|
||||
* po/zh_CN.po:
|
||||
* win32/common/_stdint.h:
|
||||
* win32/common/audio-enumtypes.c:
|
||||
* win32/common/config.h:
|
||||
* win32/common/video-enumtypes.c:
|
||||
* win32/common/video-enumtypes.h:
|
||||
Release 1.1.1
|
||||
|
||||
2013-06-05 16:20:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
||||
|
||||
|
|
|
|||
2
NEWS
2
NEWS
|
|
@ -1,2 +1,2 @@
|
|||
This is GStreamer Base Plugins 1.1.1
|
||||
This is GStreamer Base Plugins 1.1.2
|
||||
|
||||
|
|
|
|||
158
RELEASE
158
RELEASE
|
|
@ -1,5 +1,5 @@
|
|||
|
||||
Release notes for GStreamer Base Plugins 1.1.1
|
||||
Release notes for GStreamer Base Plugins 1.1.2
|
||||
|
||||
|
||||
The GStreamer team is proud to announce a new bug-fix release
|
||||
|
|
@ -63,93 +63,26 @@ Features of this release
|
|||
|
||||
Bugs fixed in this release
|
||||
|
||||
* 700342 : decodebin: Crashes and deadlocks when setting to READY while still autoplugging
|
||||
* 690197 : alsasrc: gets stuck in infinite loop if usb audio device is disconnected while being used
|
||||
* 697112 : GLTextureUploadMeta: No support for multi-texture formats
|
||||
* 634407 : decodebin should expose pads in a deterministic order
|
||||
* 636753 : pbutils function to map (container) caps to filename extension
|
||||
* 654830 : discoverer, uridecodebin, encodebin and multiple audio streams
|
||||
* 663350 : theoraenc: do not reset the encoder when we need a keyframe
|
||||
* 665751 : video: define for formats supported by gst_video_overlay_composition_blend()
|
||||
* 676884 : audiotestsrc: segment one sample too short due to rounding errors
|
||||
* 678892 : uridecodebin: differentiate between no URI handler found and URI not accepted by handler
|
||||
* 679456 : videodecoder: fix compiler optimization hint macro usage
|
||||
* 681719 : audiovisualizer does not handle VideoMeta
|
||||
* 685637 : [audioresample] Performance improvements & ARM NEON support
|
||||
* 687146 : rtpbasedepay: remove unused variable
|
||||
* 687284 : audioconvert: prefer output formats with the same depth or at least a higher depth
|
||||
* 687466 : audiobasesink: use the same type as the internal type to return it
|
||||
* 687472 : video-blend: fix memory leak
|
||||
* 687817 : textoverlay: support shaded background drawing for all formats
|
||||
* 689326 : multifdsink: document that adding fd in NULL is not allowed
|
||||
* 689845 : Encodebin API to handle multiple streams lacking
|
||||
* 690240 : encodebin: remove test of encoder name vs preset name
|
||||
* 690591 : No decoder available for type 'audio/x-avi-unknown, codec_id=(int)65534'.
|
||||
* 690994 : videodecoder: Allow parse function to not use all data on the adapter
|
||||
* 691072 : decodebin: Doesn't expose pads if no data is received before EOS
|
||||
* 692358 : appsrc deadlock setting the pipeline to NULL state
|
||||
* 692613 : tests: reduce number of wake-ups in test applications
|
||||
* 692930 : avidemux: add raw 8-bit monochrome
|
||||
* 693302 : decodebin: g_mutex_new is deprecated
|
||||
* 693401 : gstdecodebin2 doesn't set send event on pad before exposing pad
|
||||
* 693484 : uridecodebin: query URI to source element and fallback to decoder's URI
|
||||
* 693750 : Riffmedia doesn't set systemstream=false for some video/mpeg caps
|
||||
* 693862 : Crash in videoscale (with Orc enabled) on Raspberry Pi
|
||||
* 694346 : pbutils, typefinding: improve handling of MVC/SVC H.264 streams
|
||||
* 694389 : non flushing seeks after a segment done, don't sync the ringbuffer
|
||||
* 694443 : libgstaudio: add support for AAC pass-through
|
||||
* 694553 : adder: rhythmbox crossfading stopped working after commit a86ca53
|
||||
* 695203 : xvimagesink: crash in gst_xvimagesink_xvimage_put() with HLS bip-bop stream after a while
|
||||
* 695276 : libsabi test needs an update for i386
|
||||
* 695540 : riff: support raw avi with negative height
|
||||
* 695658 : build: Link libgstrtsp-1.0.so to libm for pow()
|
||||
* 695660 : appsink: update the emit-signal description
|
||||
* 695832 : audio: a print causes a floating point exception
|
||||
* 696100 : videoconvert/videoscale: broken conversion for interlaced Y41B
|
||||
* 696411 : audiotestsrc: incorrect data size in last buffer
|
||||
* 696550 : riff: add " note " tag
|
||||
* 696598 : decodebin pads no longer match order in file
|
||||
* 696818 : rtsprange: use gst_util_gdouble_to_guint64 in get_seconds
|
||||
* 696915 : decodebin: get_sticky event STREAM_START fails on newly-exposed pad
|
||||
* 696916 : videofilter doesn't add caps in pool config
|
||||
* 697628 : ximagesink: Compile error without HAVE_XSHM
|
||||
* 697631 : videoscale and videoconvert unit tests need to be updated for latest changes
|
||||
* 697665 : Add format=WMV3 for WMV 3 video
|
||||
* 697672 : VP8 passed through rtpbin decodes a single frame and then fails to decode until a key frame passed through
|
||||
* 697723 : audioringbuffer: Reset segdone when releasing audioringbuffer
|
||||
* 697808 : sdp: add boxed type for GstSDPMessage
|
||||
* 698277 : Use gst_plugin_feature_rank_compare() API instead of duplicating the code in many places
|
||||
* 698410 : Adder: Can not send flush_start and flush_stop in a row
|
||||
* 698558 : sdp: make it possible to modify session/media attributes
|
||||
* 698712 : playbin: autoplug video decoder and sink based on caps features
|
||||
* 698851 : playbin: ability to mix or play multiple audio and text streams simultaneously
|
||||
* 698888 : SDP session bandwidth not duplicated, causing segfault when freeing...
|
||||
* 699124 : vorbisdec: crash on shutdown in webkit unit test
|
||||
* 699187 : videorate: ends up outputting buffers with incorrect duration
|
||||
* 699470 : dmabuf: handle mmap failure
|
||||
* 699563 : dmabuf: fix formating
|
||||
* 699565 : dmabuf: fix memory initialization
|
||||
* 699566 : dmabuf: don't touch the GstMemory size
|
||||
* 699744 : alsasrc: timestamps provided by audiosrc subclass not used when running under slave clock
|
||||
* 699792 : oggmux: Never emitting EOS in GES
|
||||
* 699894 : videoencoder: Caps event sent before stream-start
|
||||
* 699960 : videodecoder: Reordering sticky events
|
||||
* 699971 : oggmux: Sends a segment event before sending a caps event.
|
||||
* 700006 : audio/video: base classes have suboptimal error handling when allocating a buffer not via a bufferpool
|
||||
* 700222 : rtpbasepayload: Need to delay segments event after caps event
|
||||
* 700259 : audio: fix buffer overflow for channels > 64
|
||||
* 700272 : playback: Use subset checks instead of intersections
|
||||
* 700324 : playbin hangs trying to play 4K video, and hangs again on interrupt
|
||||
* 700377 : video: add NV16 pixel format support
|
||||
* 700400 : video: can't build without orc support - implicit declaration of function 'video_orc_pack_NV16'
|
||||
* 700411 : dmabuf: Make sure that memory is unmapped before releasing it
|
||||
* 700413 : ximagesink: add alpha mask support
|
||||
* 700427 : dmabuf: set the initial memory size to the full size
|
||||
* 701202 : playsink: Badly initialized contrast/brightness
|
||||
* 701234 : SIGSEGV in videoconvert_convert_free when using fastpath
|
||||
* 701316 : rtspconnection: using g_pollable_stream_read and write breaks builds on Ubuntu and Debian stable
|
||||
* 589242 : videoconvert: need special handling for interlaced I420
|
||||
* 648359 : baseaudiosrc: ringbuffer: segbase/segdone not updated when ring buffer cleared leads to incorrect timestamps
|
||||
* 696300 : H264 video is playing too fast because of invalid PTS:
|
||||
* 698562 : rtpbuffer: broken language bindings for gst_rtp_buffer_get_payload
|
||||
* 700967 : playbin: sink messages are not received in the bus
|
||||
* 701798 : rtspsrc: Regression with connections to certain live stream
|
||||
* 701924 : tests: add playbin test for autoplugging of decoder-sink combination based on capsfeatures
|
||||
* 701943 : playbin: unit tests fail
|
||||
* 701997 : [regression] playbin: loses audio clock and hangs when switching songs after about-to-finish
|
||||
* 702195 : playbin: Emit {audio,text,video}-changed signals when pads are removed
|
||||
* 702215 : pbutils: descriptions: Allow smart codec tag handling
|
||||
* 702465 : playsink: Fix the block diagram of deinterlace bin.
|
||||
* 702467 : audiobasesrc: missing calls to gst_buffer_unmap
|
||||
* 702545 : tags: xmp: adding tags is very slow
|
||||
* 702652 : rtspconnection: tunneled connections do not work
|
||||
* 702840 : queue2 does not work with use-buffering=true
|
||||
* 703088 : rtspsrc parse error race condition
|
||||
* 703119 : rtpbasepayload leaks the segment event
|
||||
* 703196 : videoencoder: Send all pending events with type < CAPS before sending caps.
|
||||
* 703256 : typefind: Fix low probability typefinding for video/mpeg on small amount of data
|
||||
* 703477 : audioresample: disable NEON code path for 16-bit integer
|
||||
* 703684 : POTFILES.in is out of date
|
||||
|
||||
==== Download ====
|
||||
|
||||
|
|
@ -186,63 +119,20 @@ subscribe to the gstreamer-devel list.
|
|||
|
||||
Contributors to this release
|
||||
|
||||
* Akihiro Tsukada
|
||||
* Alessandro Decina
|
||||
* Alexander Schrab
|
||||
* Andoni Morales Alastruey
|
||||
* Anton Belka
|
||||
* Arnaud Vrac
|
||||
* B.Prathibha
|
||||
* Benjamin Gaignard
|
||||
* Branko Subasic
|
||||
* Brendan Long
|
||||
* Carlos Rafael Giani
|
||||
* Christian Fredrik Kalager Schaller
|
||||
* Daniel Drake
|
||||
* David Schleef
|
||||
* David Svensson Fors
|
||||
* Dirk Van Haerenborgh
|
||||
* Edward Hervey
|
||||
* Emanuele Aina
|
||||
* Evan Nemerson
|
||||
* Greg Rutz
|
||||
* Jan Schmidt
|
||||
* Jan Schole
|
||||
* Jihyun Cho
|
||||
* Jonas Holmberg
|
||||
* Jonathan Liu
|
||||
* Jose Antonio Santos Cadenas
|
||||
* Josep Torra
|
||||
* Julien Moutte
|
||||
* Marc Leeman
|
||||
* Martin Pitt
|
||||
* Matej Knopp
|
||||
* Marcin Lewandowski
|
||||
* Mathieu Duponchelle
|
||||
* Matthew Waters
|
||||
* Michael Olbrich
|
||||
* Miguel Angel Cabrera Moya
|
||||
* Nicola Murino
|
||||
* Nicolas Dufresne
|
||||
* Ognyan Tonchev
|
||||
* Olivier Crête
|
||||
* Patricia Muscalu
|
||||
* Paul HENRYS
|
||||
* Pete Beardmore
|
||||
* Philippe Normand
|
||||
* Rasmus Rohde
|
||||
* Rico Tzschichholz
|
||||
* Piotr Drąg
|
||||
* Sebastian Dröge
|
||||
* Sebastian Rasmussen
|
||||
* Simon Berg
|
||||
* Sreerenj Balachandran
|
||||
* Stefan Sauer
|
||||
* Thiago Santos
|
||||
* Thibault Saunier
|
||||
* Thijs Vermeir
|
||||
* Thomas Scheuermann
|
||||
* Tim-Philipp Müller
|
||||
* Tom Greenwood
|
||||
* Vincent Penquerc'h
|
||||
* Víctor Manuel Jáquez Leal
|
||||
* Wim Taymans
|
||||
* yanghuolin
|
||||
|
||||
|
|
@ -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],[1.1.1.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
|
||||
AC_INIT([GStreamer Base Plug-ins],[1.1.2],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
|
||||
|
||||
AG_GST_INIT
|
||||
|
||||
|
|
@ -56,10 +56,10 @@ dnl 1.2.5 => 205
|
|||
dnl 1.10.9 (who knows) => 1009
|
||||
dnl
|
||||
dnl sets GST_LT_LDFLAGS
|
||||
AS_LIBTOOL(GST, 101, 0, 101)
|
||||
AS_LIBTOOL(GST, 102, 0, 102)
|
||||
|
||||
dnl *** required versions of GStreamer stuff ***
|
||||
GST_REQ=1.1.1.1
|
||||
GST_REQ=1.1.2
|
||||
|
||||
dnl *** autotools stuff ****
|
||||
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Adds multiple streams</description>
|
||||
<filename>../../gst/adder/.libs/libgstadder.so</filename>
|
||||
<basename>libgstadder.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>ALSA plugin library</description>
|
||||
<filename>../../ext/alsa/.libs/libgstalsa.so</filename>
|
||||
<basename>libgstalsa.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Elements used to communicate with applications</description>
|
||||
<filename>../../gst/app/.libs/libgstapp.so</filename>
|
||||
<basename>libgstapp.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Convert audio to different formats</description>
|
||||
<filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename>
|
||||
<basename>libgstaudioconvert.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Adjusts audio frames</description>
|
||||
<filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename>
|
||||
<basename>libgstaudiorate.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Resamples audio</description>
|
||||
<filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename>
|
||||
<basename>libgstaudioresample.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Creates audio test signals of given frequency and volume</description>
|
||||
<filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename>
|
||||
<basename>libgstaudiotestsrc.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Read audio from CD in paranoid mode</description>
|
||||
<filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename>
|
||||
<basename>libgstcdparanoia.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>various encoding-related elements</description>
|
||||
<filename>../../gst/encoding/.libs/libgstencodebin.so</filename>
|
||||
<basename>libgstencodebin.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>GIO elements</description>
|
||||
<filename>../../gst/gio/.libs/libgstgio.so</filename>
|
||||
<basename>libgstgio.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Vorbis Tremor decoder</description>
|
||||
<filename>../../ext/vorbis/.libs/libgstivorbisdec.so</filename>
|
||||
<basename>libgstivorbisdec.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>libvisual visualization plugins</description>
|
||||
<filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
|
||||
<basename>libgstlibvisual.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>ogg stream manipulation (info about ogg: http://xiph.org)</description>
|
||||
<filename>../../ext/ogg/.libs/libgstogg.so</filename>
|
||||
<basename>libgstogg.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
@ -74,7 +74,7 @@
|
|||
<name>video_%u</name>
|
||||
<direction>sink</direction>
|
||||
<presence>request</presence>
|
||||
<details>video/x-theora; application/x-ogm-video; video/x-dirac; video/x-smoke; video/x-vp8</details>
|
||||
<details>video/x-theora; application/x-ogm-video; video/x-dirac; video/x-smoke; video/x-vp8; video/x-daala</details>
|
||||
</caps>
|
||||
<caps>
|
||||
<name>src</name>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Pango-based text rendering and overlay</description>
|
||||
<filename>../../ext/pango/.libs/libgstpango.so</filename>
|
||||
<basename>libgstpango.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>various playback elements</description>
|
||||
<filename>../../gst/playback/.libs/libgstplayback.so</filename>
|
||||
<basename>libgstplayback.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Subtitle parsing</description>
|
||||
<filename>../../gst/subparse/.libs/libgstsubparse.so</filename>
|
||||
<basename>libgstsubparse.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>transfer data over the network via TCP</description>
|
||||
<filename>../../gst/tcp/.libs/libgsttcp.so</filename>
|
||||
<basename>libgsttcp.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Theora plugin library</description>
|
||||
<filename>../../ext/theora/.libs/libgsttheora.so</filename>
|
||||
<basename>libgsttheora.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>default typefind functions</description>
|
||||
<filename>../../gst/typefind/.libs/libgsttypefindfunctions.so</filename>
|
||||
<basename>libgsttypefindfunctions.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Colorspace conversion</description>
|
||||
<filename>../../gst/videoconvert/.libs/libgstvideoconvert.so</filename>
|
||||
<basename>libgstvideoconvert.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Adjusts video frames</description>
|
||||
<filename>../../gst/videorate/.libs/libgstvideorate.so</filename>
|
||||
<basename>libgstvideorate.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Resizes video</description>
|
||||
<filename>../../gst/videoscale/.libs/libgstvideoscale.so</filename>
|
||||
<basename>libgstvideoscale.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Creates a test video stream</description>
|
||||
<filename>../../gst/videotestsrc/.libs/libgstvideotestsrc.so</filename>
|
||||
<basename>libgstvideotestsrc.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>plugin for controlling audio volume</description>
|
||||
<filename>../../gst/volume/.libs/libgstvolume.so</filename>
|
||||
<basename>libgstvolume.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>Vorbis plugin library</description>
|
||||
<filename>../../ext/vorbis/.libs/libgstvorbis.so</filename>
|
||||
<basename>libgstvorbis.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>X11 video output element based on standard Xlib calls</description>
|
||||
<filename>../../sys/ximage/.libs/libgstximagesink.so</filename>
|
||||
<basename>libgstximagesink.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -3,7 +3,7 @@
|
|||
<description>XFree86 video output plugin using Xv extension</description>
|
||||
<filename>../../sys/xvimage/.libs/libgstxvimagesink.so</filename>
|
||||
<basename>libgstxvimagesink.so</basename>
|
||||
<version>1.1.1</version>
|
||||
<version>1.1.2</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>GStreamer Base Plug-ins source release</package>
|
||||
|
|
|
|||
|
|
@ -34,6 +34,16 @@ A wide range of video and audio decoders, encoders, and filters are included.
|
|||
</GitRepository>
|
||||
</repository>
|
||||
|
||||
<release>
|
||||
<Version>
|
||||
<revision>1.1.2</revision>
|
||||
<branch>1.1</branch>
|
||||
<name></name>
|
||||
<created>2013-07-11</created>
|
||||
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.1.2.tar.xz" />
|
||||
</Version>
|
||||
</release>
|
||||
|
||||
<release>
|
||||
<Version>
|
||||
<revision>1.1.1</revision>
|
||||
|
|
|
|||
|
|
@ -1,8 +1,8 @@
|
|||
#ifndef _GST_PLUGINS_BASE__STDINT_H
|
||||
#define _GST_PLUGINS_BASE__STDINT_H 1
|
||||
#ifndef _GENERATED_STDINT_H
|
||||
#define _GENERATED_STDINT_H "gst-plugins-base 1.1.1"
|
||||
/* generated using gnu compiler gcc-4.8 (Debian 4.8.1-2) 4.8.1 */
|
||||
#define _GENERATED_STDINT_H "gst-plugins-base 1.1.2"
|
||||
/* generated using gnu compiler gcc-4.8 (Debian 4.8.1-6) 4.8.1 */
|
||||
#define _STDINT_HAVE_STDINT_H 1
|
||||
#include <stdint.h>
|
||||
#endif
|
||||
|
|
|
|||
|
|
@ -87,7 +87,7 @@
|
|||
#define GST_PACKAGE_ORIGIN "Unknown package origin"
|
||||
|
||||
/* GStreamer package release date/time for plugins as YYYY-MM-DD */
|
||||
#define GST_PACKAGE_RELEASE_DATETIME "2013-06-05"
|
||||
#define GST_PACKAGE_RELEASE_DATETIME "2013-07-11"
|
||||
|
||||
/* Define if static plugins should be built */
|
||||
#undef GST_PLUGIN_BUILD_STATIC
|
||||
|
|
@ -328,7 +328,7 @@
|
|||
#define PACKAGE_NAME "GStreamer Base Plug-ins"
|
||||
|
||||
/* Define to the full name and version of this package. */
|
||||
#define PACKAGE_STRING "GStreamer Base Plug-ins 1.1.1"
|
||||
#define PACKAGE_STRING "GStreamer Base Plug-ins 1.1.2"
|
||||
|
||||
/* Define to the one symbol short name of this package. */
|
||||
#define PACKAGE_TARNAME "gst-plugins-base"
|
||||
|
|
@ -337,7 +337,7 @@
|
|||
#undef PACKAGE_URL
|
||||
|
||||
/* Define to the version of this package. */
|
||||
#define PACKAGE_VERSION "1.1.1"
|
||||
#define PACKAGE_VERSION "1.1.2"
|
||||
|
||||
/* directory where plugins are located */
|
||||
#ifdef _DEBUG
|
||||
|
|
@ -371,7 +371,7 @@
|
|||
#undef USE_TREMOLO
|
||||
|
||||
/* Version number of package */
|
||||
#define VERSION "1.1.1"
|
||||
#define VERSION "1.1.2"
|
||||
|
||||
/* Define WORDS_BIGENDIAN to 1 if your processor stores words with the most
|
||||
significant byte first (like Motorola and SPARC, unlike Intel). */
|
||||
|
|
|
|||
Loading…
Reference in a new issue