diff options
36 files changed, 1470 insertions, 85 deletions
@@ -1,9 +1,1255 @@ +=== release 1.5.90 === + +2015-08-19 Sebastian Dröge <slomo@coaxion.net> + + * configure.ac: + releasing 1.5.90 + +2015-08-19 11:23:09 +0300 Sebastian Dröge <sebastian@centricular.com> + + * po/cs.po: + * po/da.po: + * po/de.po: + * po/hu.po: + * po/nb.po: + * po/pl.po: + * po/ru.po: + * po/uk.po: + * po/zh_CN.po: + po: Update translations + +2015-08-19 08:37:46 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * tools/gst-discoverer.c: + tools: discoverer: When info is NULL just print error and return + In case discover_uri returns NULL info, passing the info to discoverer APIs + result in critical assertion errors. Hence instead of passing NULL info along, + print the error and return. + https://bugzilla.gnome.org/show_bug.cgi?id=753701 + +2015-08-18 18:47:22 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + Revert "decodebin: Handle the preroll multi-queue size" + This reverts commit 5c8ef0ea05123506dfc35c70c8b165bca7435dad. + +2015-08-18 18:47:21 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + Revert "decodebin: Store extra_buffer_required per group, not globally" + This reverts commit 1ea81114ea6bd48b581f19002018680933aa7a12. + +2015-08-18 18:47:18 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + Revert "decodebin: If extra buffers are going to be required, we're still prerolling" + This reverts commit a3b24f0241bd55a005a072ba8ddcd53e0fdbf827. + +2015-08-18 16:28:42 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideodecoder.c: + * gst-libs/gst/video/gstvideoencoder.c: + video(en|de)coder: Return TRUE when we consumed a tag event without creating a new event + Fixes spurious flow errors that especially break gst-validate. + +2015-08-18 16:01:28 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: If there are no tags, don't try to do event handling on a NULL event + Fixes some crashes. + +2015-08-18 15:58:57 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: If there are no tags, don't try to do event handling on a NULL event + Fixes some crashes. + +2015-08-18 13:50:17 +0300 Vivia Nikolaidou <vivia@ahiru.eu> + + * tools/gst-play.c: + tools: gst-play: Use g_build_filename instead of g_strconcat + When running gst-play against a directory name, and suffix the path with a + directory separator (e.g. tab completion), gst-play was printing two directory + separators in a row. g_build_filename fixes this, and additionally allows for + both '/' and '\' as separators on Windows. + +2015-08-18 15:16:25 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: If extra buffers are going to be required, we're still prerolling + +2015-08-18 15:01:33 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: Store extra_buffer_required per group, not globally + It's only relevant for each group, and by storing it in the group + we have locking and everything else like for the other buffering-related + variables. Locking looks a bit fishy still, but it was like that for a long + time already so shouldn't be worse than before. + +2015-07-30 10:33:25 +0900 Myoungsun Lee <ohmygod0327@gmail.com> + + * gst/playback/gstdecodebin2.c: + decodebin: Handle the preroll multi-queue size + Overview: + There are some of interleaved streams which has long-term location of audio data. + It mean the audio data is located far away more than multiqueue size. + In this case, because of multiqueue overrun, the pipeline is stopped. + To prevent hanging-like state, the decodebin needs to handle the queue size. + Caused: + The multiqueue size is not enough, the pipeline will stay being stalled status + and decodebin cannot complete to build decode chain. + In this issue file, decodebin did not receive no_more_pads signal or audio data yet. + Steps to Reproduce: + play the high-resolution(4K file) files or some streaming media(push mode). + Actual Results: + There is no audio or subtitle. + We can see only video or infinite loading. + Resolution: + Decodebin detect this problem, and add extra buffer size to multiqueue. + The multiqueue is larger than before, the next data can be pushed the downstream element. + Additional Information: + The max-preroll extra buffer size is set 8MB. + We can use total pre-roll buffer 10MB. + Only first overrun callback can handle multiqueue size. + https://bugzilla.gnome.org/show_bug.cgi?id=733235 + +2015-08-18 12:29:29 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/video/gstvideoencoder.c: + videoencoder: fix tag handling + Merge upstream tags with encoder tags and update whenever + any of those changes. + https://bugzilla.gnome.org/show_bug.cgi?id=679768 + +2015-08-18 11:45:24 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: fix tag handling + Merge upstream tags with encoder tags and update whenever + any of those changes. + https://bugzilla.gnome.org/show_bug.cgi?id=679768 + +2015-08-18 12:56:33 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/typefind/gsttypefindfunctions.c: + typefindfunctions: Add typefinder for TTML+XML + Used in DASH among other things, as SMPTE Timed Text. + +2015-08-18 09:06:39 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * gst-libs/gst/pbutils/gstdiscoverer.c: + pbutils: discoverer: Set GError when NULL info is being returned. + When discovering the URI, if info is NULL, then instead of just returning NULL, + set the GError, so the error can be printed and notified. + https://bugzilla.gnome.org/show_bug.cgi?id=753701 + +2015-08-17 11:18:25 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * tools/gst-discoverer.c: + discoverer: free context and error during failures + When g_option_context_parse or gst_discoverer_new fails, then there will + be memory leaks for ctx and err variables. Free'ing the same. + https://bugzilla.gnome.org/show_bug.cgi?id=753701 + +2015-08-16 18:28:09 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: try harder to avoid sending unnecessary tag updates + +2015-08-16 17:55:22 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: fix tag handling + Before we just merged everything in pretty much random ways + ad-hoc instead of keeping state properly. In 0.10 that was + how it worked, but in 1.x the tag events sent should always + reflect the latest state and replace any previous tags. + So save the upstream (stream) tags, and save the tags set + by the decoder subclass with merge mode, and then update + the merged tags whenever either of those two changes. + This slightly changes the behaviour of gst_video_decoder_merge_tags() + in case it is called multiple times, since now any call replaces + the previously-set tags. However, it leads to much more predictable + outcomes, and also we are not aware of any subclass which sets this + multiple times and expects all the tags set to be merged. + If more complex tag merging scenarios are required, we'll have + to add a new vfunc for that or the subclass has to intercept + the upstream tags itself and send merged tags itself. + https://bugzilla.gnome.org/show_bug.cgi?id=679768 + +2015-08-14 17:59:29 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/libs/audiodecoder.c: + tests: audiodecoder: add unit test for tag handling + https://bugzilla.gnome.org/show_bug.cgi?id=679768 + +2015-08-14 17:44:59 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: fix tag handling + Before we just merged everything in pretty much random ways + ad-hoc instead of keeping state properly. In 0.10 that was + how it worked, but in 1.x the tag events sent should always + reflect the latest state and replace any previous tags. + So save the upstream (stream) tags, and save the tags set + by the decoder subclass with merge mode, and then update + the merged tags whenever either of those two changes. + This slightly changes the behaviour of gst_audio_decoder_merge_tags() + in case it is called multiple times, since now any call replaces + the previously-set tags. However, it leads to much more predictable + outcomes, and also we are not aware of any subclass which sets this + multiple times and expects all the tags set to be merged. + If more complex tag merging scenarios are required, we'll have + to add a new vfunc for that or the subclass has to intercept + the upstream tags itself and send merged tags itself. + https://bugzilla.gnome.org/show_bug.cgi?id=679768 + +2015-08-15 22:23:15 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/vorbis/gstvorbisenc.c: + vorbisenc: use template subset check for accept-caps + It is faster than doing a query that propagates downstream and + should be enough + +2015-08-16 12:20:51 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/vorbis/gstvorbisenc.c: + vorbisenc: use more accurate sink pad template caps + Removes the need for custom caps query handling and makes it more + correct from the beginning on the template. It is a bit uglier + to read because there is 1 entry per channel but makes code easier + to maintain. + +2015-08-15 22:22:41 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/theora/gsttheoraenc.c: + theoraenc: use template subset check for accept-caps + It is faster than doing a query that propagates downstream and + should be enough + +2015-08-16 08:12:01 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/audio/gstaudioencoder.h: + audioencoder: add src and sink query methods + Allows subclasses to do their own handling of GstQuery and still + chain up to the parent class to handle the ones that they don't want + to handle + +2015-08-16 12:53:02 +0200 Edward Hervey <bilboed@bilboed.com> + + * gst/playback/gstdecodebin2.c: + decodebin: Fix list iteration + We were using the wrong variable ... + CID #1316477 + +2015-05-04 11:19:28 +0200 Edward Hervey <edward@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin2: Handle flushing with multiple decode groups + When an upstream element wants to flush downstream, we need to take + all chains/groups into consideration. + To that effect, when a FLUSH_START event is seen, after having it + sent downstream we mark all those chains/groups as "drained" (as if + they had seen a EOS event on the endpads). + When a FLUSH_STOP event is received, we check if we need to switch groups. + This is done by checking if there are next groups. If so, we will switch + over to the latest next_group. The actual switch will be done when + that group is blocked. + https://bugzilla.gnome.org/show_bug.cgi?id=606382 + +2015-04-29 15:56:39 +0200 Edward Hervey <edward@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin2: Forward event/queries for unlinked groups + When upstream events/queries reach sinkpads of unlinked groups (i.e. + no longer linked to the upstream demuxer), this patch attempts to find + the linked group and forward it upstream of that group. + This is done by adding upstream event/query probes on new group sinkpads + and then: + * Checking if the pad is linked or not (has a peer or not) + * If there is a peer, just let the event/query follow through normally + * If there is no peer, we find a pad to which to proxy it and return + GST_PROBE_HANDLED if it succeeded (allowing the event/query to be properly + returned to the initial called) + Note that this is definitely not thread-safe for the time being + https://bugzilla.gnome.org/show_bug.cgi?id=606382 + +2015-08-15 08:18:59 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + * win32/common/libgstaudio.def: + Revert "audiodecoder: expose default query handling function" + Apparently I forgot how gobject works, there is no need to expose + it directly as one can call it from the parent_class pointer + This reverts commit 8a64592481dab985ca520a5b1cb394a609275c60. + +2015-08-15 08:14:00 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + * gst-libs/gst/video/gstvideodecoder.h: + * win32/common/libgstvideo.def: + Revert "videodecoder: expose default query handling function" + Apparently I forgot how gobject works, there is no need to expose + it directly as one can call it from the parent_class pointer + This reverts commit ea9b6a7e3c4eea512650adf530b7f1acb0eccd84. + +2015-08-15 07:41:24 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/vorbis/gstvorbisdec.c: + vorbisdec: use default pad accept-caps handling + Avoids useless check of downstream caps when handling an + accept-caps query + +2015-08-15 07:40:55 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * ext/theora/gsttheoradec.c: + theoradec: use default pad accept-caps handling + Avoids useless check of downstream caps when handling an + accept-caps query + +2015-08-15 07:31:54 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + * win32/common/libgstaudio.def: + audiodecoder: add option to use default pad accept-caps handling + Add gst_audio_decoder_set_use_default_pad_acceptcaps() to allow + subclasses to make videodecoder use the default pad acceptcaps + handling instead of resorting to the caps query that is, usually, + less efficient and unecessary + API: gst_audio_decoder_set_use_default_pad_acceptcaps + +2015-08-15 07:20:25 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + * gst-libs/gst/video/gstvideodecoder.h: + * win32/common/libgstvideo.def: + videodecoder: add option to use default pad accept-caps handling + Add gst_video_decoder_set_use_default_pad_acceptcaps() to allow + subclasses to make videodecoder use the default pad acceptcaps + handling instead of resorting to the caps query that is, usually, + less efficient and unecessary + API: gst_video_decoder_set_use_default_pad_acceptcaps + +2015-08-15 23:33:14 +1000 Jan Schmidt <jan@centricular.com> + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + rtpbasedepayload: Make stats creation threadsafe, fix a CRITICAL + Use the object lock to protect the internal segment when updating + against access from getting the stats property. + Fix a critical in gst-inspect or when retrieving the stats + before any segment has arrived by checking whether the + segment has been initted.. + +2015-08-12 03:00:15 +1000 Jan Schmidt <jan@centricular.com> + + * gst/typefind/gsttypefindfunctions.c: + typefind: Make the H.264 typefind a tiny bit more lenient. + When we see prefix NALs before a Subset SPS has been spotted, + it might just be because the stream was truncated at the + start, so don't count those as either 'bad' or 'good' packets. + +2015-08-14 18:43:03 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com> + + * gst-libs/gst/app/gstappsink.c: + appsink: unref the preroll buffer and cleanup the segments on stop() + Just for consistency. No need to keep data around. + +2015-08-14 18:35:22 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com> + + * gst-libs/gst/app/gstappsink.c: + appsink: do not update preroll_caps unless the sink is prerolling + Just for consistency with the preroll_segment + +2015-08-14 18:06:03 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com> + + * tests/check/elements/appsink.c: + tests/appsink: add test to ensure that the segment returned by pull-preroll/sample is correct + https://bugzilla.gnome.org/show_bug.cgi?id=751147 + +2015-06-18 12:30:24 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com> + + * gst-libs/gst/app/gstappsink.c: + appsink: put the correct segment in the preroll sample + last_segment is only being updated in dequeue_buffer(), + which is only called from _pull_sample(). _pull_preroll() + simply re-uses an old or dummy segment while the actual + one sits and waits in the queue. + https://bugzilla.gnome.org/show_bug.cgi?id=751147 + +2015-08-14 08:59:51 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + * gst-libs/gst/video/gstvideodecoder.h: + * win32/common/libgstvideo.def: + videodecoder: expose default query handling function + Subclasses can use it to select what queries they want to handle + and forward the rest to the default handling function. + API: gst_video_decoder_sink_query_default + https://bugzilla.gnome.org/show_bug.cgi?id=753623 + +2015-08-14 08:58:58 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + * win32/common/libgstaudio.def: + audiodecoder: expose default query handling function + Subclasses can use it to select what queries they want to handle + and forward the rest to the default handling function. + API: gst_audio_decoder_sink_query_default + https://bugzilla.gnome.org/show_bug.cgi?id=753623 + +2015-08-14 11:11:10 +0200 Edward Hervey <bilboed@bilboed.com> + + * tests/check/generic/states.c: + check: Rename states unit test + Makes it easier to differentiate from other modules states unit test + +2015-08-14 05:48:31 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: remove accept-caps handling + Just let the internal element of the bin do it instead of forcing a + caps query to do it. + +2015-08-13 13:52:17 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/videorate/gstvideorate.c: + videorate: fixate the pixel-aspect-ratio + If the pixel-aspect-ratio is not fixed, try to get it as close + to 1/1 as possible + https://bugzilla.gnome.org/show_bug.cgi?id=748635 + +2015-08-11 15:09:10 +0100 Tim-Philipp Müller <tim@centricular.com> + + * ext/theora/gsttheoraenc.c: + theoraenc: mention videorate is often needed in docs + https://bugzilla.gnome.org/show_bug.cgi?id=748877 + +2015-08-11 14:10:57 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/Makefile.am: + rtp: Depend on the audio library + +2015-07-01 16:25:13 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/rtp/gstrtpbaseaudiopayload.c: + rtpbaseaudiopayload: Copy metadata in the (de)payloader, but only the relevant ones + The payloader didn't copy anything so far, the depayloader copied every + possible meta. Let's make it consistent and just copy all metas without + tags or with only the audio tag. + https://bugzilla.gnome.org/show_bug.cgi?id=751774 + +2015-08-10 22:03:48 +0200 Joan Pau Beltran <joanpau.beltran@socib.cat> + + * gst/videorate/gstvideorate.c: + videorate: add support for bayer formats + Since the videorate element just duplicates or drops frames + to achieve the desired framerate, it can accept video/x-bayer media + (in any format), which are not present in the current caps. + Just add "video/x-bayer(ANY);" to the caps of the static pad template + (fixing line style to pass the indent commit hook). + https://bugzilla.gnome.org/show_bug.cgi?id=753483 + +2015-08-05 15:32:54 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + basedepayloader: Don't re-timestamp with running-time + There was a confusion, six depayloaders where passing through the + timestamp while the base class was re-timestamping to running + time. This inconstancy has been unnoticed has in most use cases + the incoming segment is [0, inifnity] in which case timestamps are + the same as running time. With DTS/PTS shifting added (to avoid + negative values) and pcapparse sending a different segment this + started being an issue. + https://bugzilla.gnome.org/show_bug.cgi?id=753037 + +2015-08-10 09:49:19 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + videoencoder: remove empty line to make g-i-scanner happy + gstvideoencoder.h:228: Warning: GstVideo: "@transform_meta" + parameter unexpected at this location: + * @transform_meta: Optional. Transform the metadata on ... + +2015-08-10 08:17:09 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: documentation cleanup + Remove some whitespace and break lines longer than 80 columns + +2015-08-10 00:21:42 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * tests/check/libs/audiodecoder.c: + tests: audiodecoder: add test to make sure gap is pushed before segment + https://bugzilla.gnome.org/show_bug.cgi?id=753360 + +2015-08-09 23:23:05 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + * tests/check/libs/videodecoder.c: + videodecoder: push pending events before gap + Push all pending events before pushing the gap. This ensures the + segment is pushed before the gap so it can be properly translated + to the running time + Includes unit test. + https://bugzilla.gnome.org/show_bug.cgi?id=753360 + +2015-07-30 16:39:03 -0400 Olivier Crête <olivier.crete@collabora.com> + + * ext/ogg/gstoggdemux.c: + oggdemux: Set chain pointers to NULL + Otherwise, they will refer to freed memory + https://bugzilla.gnome.org/show_bug.cgi?id=753078 + +2015-07-31 13:31:56 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * gst/playback/gstdecodebin2.c: + decodebin: fix deadend_details string leak + deadend_details need not be returned when the pad is not a deadend. + Hence checking if res value is TRUE and clearing the string instead of + passing it on + https://bugzilla.gnome.org/show_bug.cgi?id=753088 + +2015-08-04 14:41:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * gst/videotestsrc/gstvideotestsrc.c: + videotestsrc: Don't set DTS on buffer + DTS is for encoded data and have no meaning for raw. It better to not + set it, as it's confusing. + https://bugzilla.gnome.org/show_bug.cgi?id=752791 + +2015-07-30 18:43:19 -0400 Olivier Crête <olivier.crete@collabora.com> + + * ext/ogg/gstoggdemux.c: + oggdemux: Return FLUSHING if pad if flushing + If the initial seek fails because the pad is + flushing, then return GST_FLOW_FLUSHING instead + of an error. + +2015-07-30 15:16:57 +0100 Brian Peters <brianfpeters@gmail.com> + + * gst-libs/gst/rtp/gstrtpbuffer.c: + rtpbuffer: avoid accessing NULL buffer even more + Previous commit was incompletely applied. + https://bugzilla.gnome.org/show_bug.cgi?id=753001 + +2015-07-30 14:30:44 +0100 Brian Peters <brianfpeters@gmail.com> + + * gst-libs/gst/rtp/gstrtpbuffer.c: + rtp: buffer: don't access NULL buffer pointer + unmap will set rtpbuffer->buffer to NULL, so we need to + save the pointer to access it while the RTP buffer is + unmapped. + https://bugzilla.gnome.org/show_bug.cgi?id=753001 + +2015-07-30 12:50:56 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + rtpbasedepayload: fix leaks in error code paths + This was introduced when reshuffling the buffer unmaps + in commit bc14cdf529e21356ea7b2c8f34614958a91f7260 + rtp: rtpbasedepayload: add process_rtp_packet() vfunc + Fixes make check-valgrind. + https://bugzilla.gnome.org/show_bug.cgi?id=750235 + +2015-07-28 13:57:20 +0300 Sebastian Dröge <sebastian@centricular.com> + + * ext/pango/gstbasetextoverlay.c: + textoverlay: Query downstream caps for checking if caps features are supported, not just accept-caps + accept-caps is not recursive and might stop at the next downstream element, + while caps queries are generally recursive. The next element might accept any + capsfeatures we want, but that doesn't mean that further downstream it will + also work. + Additionally for the future: + We should probably check if downstream *prefers* the + overlay meta, and only enforce usage of it if we can't handle + the format ourselves and thus would have to drop the overlays. + Otherwise we should prefer what downstream wants here. + +2015-07-23 15:28:42 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk> + + * ext/pango/gstbasetextoverlay.c: + * ext/pango/gstbasetextoverlay.h: + basetextoverlay: Use the extents rectangle for positioning + the extents rectangle is what you need to know to properly position + a buffer that has been rendered in a surface of the ink rectangle + size. This patch make the placement on par with the placement we had + before without having to over allocate. + This patch also enable placement for vertical rendering. Note that + the halginement, valighment and line-alignment default are set to + the previous default when this property is set. This is for backward + compatibility, you can change the value after setting vertical render. + https://bugzilla.gnome.org/show_bug.cgi?id=728636 + +2015-07-23 15:19:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk> + + * ext/pango/gstbasetextoverlay.c: + basetextoverlay: Fix clipping issues + This patch uses the ink rectangle in order to compute the size + of the surface require to render. It also correctly compute the + transformation matrix as the ink_rect position might not be at + 0, 0. Additionally, shadow_offset and outline_offset (which is + in fact the diameter of a dot, not a really an offset) is now + taken into account. Redundant matrix operation has been removed + for the vertical rendering. + Take note that the matrix operation in cairo are excuted in + reverse order. + https://bugzilla.gnome.org/show_bug.cgi?id=728636 + +2015-07-24 10:15:21 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-play.c: + tools: gst-play: seek at least in steps of a second + In case of very short files we might end up seeking in + steps of a fraction of a second, which is silly and gives + the impression that seeking doesn't actually work. Make + minimum seek step a second instead. + +2015-07-22 16:19:48 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * ext/pango/gstbasetextoverlay.c: + basetextoverlay: Improve further the negotiation function + * Only send the caps event once if the query had support for the + overlay composition meta. + * Only do the allocation query if it is supported through caps. + * Send overlay_caps before doing allocation query rather then normal + caps + https://bugzilla.gnome.org/show_bug.cgi?id=751157 + +2015-07-22 20:50:10 +0200 Rico Tzschichholz <ricotz@ubuntu.com> + + * ext/pango/Makefile.am: + basetextoverlay: Add missing linking against -lm + +2015-07-21 18:40:59 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * ext/pango/gstbasetextoverlay.c: + * ext/pango/gstbasetextoverlay.h: + basetextoverlay: Ensure meta coordinate are in stream scale + The GstVideoOverlayComposition meta coordinates should always be + in stream scale, regardless of the window size downstream. This + way the sink can always scale the composition if the window size + have changed after a buffer (with his meta) was rendered before. + https://bugzilla.gnome.org/show_bug.cgi?id=751157 + +2015-07-21 14:12:41 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * ext/pango/gstbasetextoverlay.c: + * ext/pango/gstbasetextoverlay.h: + basetextoverlay: Reorder and cleanup class attribute + Also add a minimum amount of comment so we can understand what + is doing what. + https://bugzilla.gnome.org/show_bug.cgi?id=751157 + +2015-07-15 21:56:17 +0300 Ville Skyttä <ville.skytta@iki.fi> + + * gst/typefind/gsttypefindfunctions.c: + typefind: Treat *.umx (Unreal Music Package) as audio/x-mod + https://bugzilla.gnome.org//show_bug.cgi?id=752436 + +2015-07-20 16:25:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk> + + * ext/pango/gstbasetextoverlay.c: + basetextoverlay: Fix upstream composition handling + We need to update the render when upstream composition changes + or if it was removed. + http://bugzilla.gnome.org/show_bug.cgi?id=751157 + +2015-07-20 16:20:24 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk> + + * ext/pango/gstbasetextoverlay.c: + basetextoverlay: Clear reconfigure flags before negotation + This avoids negotiating twice. Current the _setcaps() patch does + not clear the initial reconfigure flags, which lead to systematic + double renegotiation. + http://bugzilla.gnome.org/show_bug.cgi?id=751157 + +2015-07-20 15:55:07 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk> + + * ext/pango/gstbasetextoverlay.c: + basetestoverlay: Always query window dimension + Remove the optimization to skip allocation query so we can + always have the latest window size information. Also, correctly + deal with the case where there is no window size information. + http://bugzilla.gnome.org/show_bug.cgi?id=751157 + +2015-07-20 15:11:06 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk> + + * ext/pango/gstbasetextoverlay.c: + basetextoverlay: Send caps before doing allocation query + This is currently a limitation of BaseTransform base class. Which means + pretty much every filters out there. + http://bugzilla.gnome.org/show_bug.cgi?id=751157 + +2015-06-18 06:31:00 +0200 Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk> + + * ext/pango/gstbasetextoverlay.c: + basetextoverlay: Log GstVideoOverlayComposition negotiation + https://bugzilla.gnome.org/show_bug.cgi?id=751157 + +2015-03-25 14:10:10 +0100 Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk> + + * ext/pango/gstbasetextoverlay.c: + * ext/pango/gstbasetextoverlay.h: + basetextoverlay: Receive window size event and adjust rendering + * cache window size event and update handle ratio + * init width with 1, don't use 0 + * don't update overlay when receiving same window size + * receive window size from allocation query + https://bugzilla.gnome.org/show_bug.cgi?id=751157 + +2015-03-19 17:59:16 +0100 Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk> + + * ext/pango/gstbasetextoverlay.c: + * ext/pango/gstbasetextoverlay.h: + basetestoverlay: Pass down meta buffers from upstream that supports GstVideoOverlayComposition + This makes pipelines with multiple textoverlay elements possible. + The meta data is collected from the upstream textoverlay element, + merged into a new GstVideoOverlayComposition and passed down downstream. + https://bugzilla.gnome.org/show_bug.cgi?id=751157 + +2015-07-10 12:49:01 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk> + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + depayloader: Use input segment start + When there is no clock_base provided, the start position is + set to 0 instead of the original segment start value. This + would break synchronization if start was not 0. + https://bugzilla.gnome.org/show_bug.cgi?id=752228 + +2015-07-16 21:26:30 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/typefind/gsttypefindfunctions.c: + typefindfunctions: add DASH MPD typefinder + Moved from dashdemux plugin in -bad. + +2015-07-16 10:07:45 +0900 Vineeth T M <vineeth.tm@samsung.com> + + * tests/examples/seek/jsseek.c: + jsseek: fix memory leaks + ctx, list and visual_entries are not being freed + resulting in memory leaks + https://bugzilla.gnome.org/show_bug.cgi?id=752454 + +2015-07-16 17:15:33 +0100 Tim-Philipp Müller <tim@centricular.com> + + * ext/ogg/gstogmparse.c: + * ext/pango/gsttextrender.c: + * gst/subparse/gstsubparse.c: + * gst/videoconvert/gstvideoconvert.c: + Update mailing list address from sourceforge to freedesktop + +2015-07-16 10:54:29 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-device-monitor.c: + tools: gst-device-monitor: fix props leak + CID 1311942 + +2015-07-15 18:22:28 +0200 Wim Taymans <wtaymans@redhat.com> + + * tools/gst-device-monitor.c: + device-monitor: print device properties + +2015-07-15 12:45:10 +0200 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/gstvideometa.c: + * gst-libs/gst/video/gstvideopool.c: + * gst-libs/gst/video/video-chroma.c: + * gst-libs/gst/video/video-color.c: + * gst-libs/gst/video/video-converter.c: + * gst-libs/gst/video/video-info.c: + * gst-libs/gst/video/video-resampler.c: + * gst-libs/gst/video/video-scaler.c: + * gst-libs/gst/video/videooverlay.c: + * gst/videoscale/gstvideoscale.c: + * gst/videotestsrc/videotestsrc.c: + video: improve logging + Add logging categories for most video objects. + Remove some useless debug lines in video-info and videotestsrc. + Add a performance debug line in the video scaler. + +2015-07-15 12:46:07 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * tests/examples/seek/jsseek.c: + jsseek: fix tag list leak + tags are being leaked while updating the streams in jsseek + https://bugzilla.gnome.org/show_bug.cgi?id=752400 + +2015-07-15 10:50:46 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * tests/examples/playback/playback-test.c: + playback-test: fix tag list leak + tags are being leaked while updating the streams in playback-test + https://bugzilla.gnome.org/show_bug.cgi?id=752397 + +2015-07-14 17:17:34 -0400 Olivier Crête <olivier.crete@collabora.com> + + * gst-libs/gst/rtsp/gstrtsptransport.h: + rtsp: Include generated enum types in gstrtsptransport.h + GST_TYPE_RTSP_LOWER_TRANS used to be defined in there, not + including the generated file makes older gst-p-good fail to build, + so it constitues an API break. + +2015-07-14 15:58:43 +0200 Wim Taymans <wtaymans@redhat.com> + + * gst/tcp/gstsocketsrc.c: + * gst/tcp/gstsocketsrc.h: + socketsrc: add caps property + Add caps property that allows the src to easily negotiate a format. + +2015-07-14 13:00:03 +0900 Vineeth T M <vineeth.tm@samsung.com> + + * tests/examples/playback/playback-test.c: + playback-test: fix memory leak + context during main and filter list during init + visualization are not being freed resulting in memory leak + and app->vis_entries + https://bugzilla.gnome.org/show_bug.cgi?id=752359 + +2015-07-14 00:03:10 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/playback/gstdecodebin2.c: + decodebin: only try to expose complete groups + When switching to a new chain it might be that this new chain + is not yet ready to be exposed so check it before exposing. + Can happen with mpegts that might delay adding pads or pushing data + until it has found the PMT/PAT/PCR and that may take a while depending + on the stream. + It happened frequently with HLS: + http://vevoplaylist-live.hls.adaptive.level3.net/vevo/ch1/appleman.m3u8 + +2015-07-14 00:02:40 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/playback/gstdecodebin2.c: + decodebin: fix typo + Hided -> hid + +2015-05-27 18:55:20 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + * gst-libs/gst/rtp/gstrtpbasedepayload.h: + rtp: rtpbasedepayload: add process_rtp_packet() vfunc + Add process_rtp_packet() vfunc that works just like the + existing process() vfunc only that it takes the GstRTPBuffer + that the base class has already mapped (with MAP_READ), + which means that the subclass doesn't have to map it again, + which allows more performant processing of input buffers + for most RTP depayloaders. + https://bugzilla.gnome.org/show_bug.cgi?id=750235 + +2015-07-10 11:53:24 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplaysink.c: + playsink: Require the streamvolume interface on the sink when using the sink's volume/mute properties + If the sink has properties named volume and mute, we have no idea about their + meaning. The streamvolume interface standardizes the meaning. + In the case of osxaudiosink for example, the current volume property has a + range of 0.0 to 1.0, but we need 0.0 to 10.0 or similar. Also osxaudiosink + has no mute property. As such, the volume element should be used here instead. + https://bugzilla.gnome.org/show_bug.cgi?id=752156 + +2015-07-09 10:47:20 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk> + + * gst-libs/gst/video/video-frame.h: + doc/build: Fix doc typos + This minor update should workaround a build system bug. While the + makefile has been updated to generate more enum type, there is nothing + that updates the header and would lead to the generated code to be + produced again. This minor doc fix should ensure no one get a build with + missing symbols. + +2015-07-09 17:20:55 +0300 Sebastian Dröge <sebastian@centricular.com> + + * win32/common/libgstvideo.def: + Revert "win32 def: Remove video flags symbol that don't exist" + This reverts commit b20cc6a02a007521eabceeceb60356e5a252f38a. + They are actually there in the autogenerated enum header/source file. + +2015-07-09 10:15:11 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk> + + * win32/common/libgstvideo.def: + win32 def: Remove video flags symbol that don't exist + There has been a some refactoring and these symbols don't exist anynmore. + So remove it from the win32 def. This should fix distcheck. + +2015-07-07 19:56:52 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + rtpbasedepayload: fix typo in comment + +2015-07-07 15:05:59 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + rtpbasepayload: fix possible segment event leak + Need to clear it when shutting down, not when starting up. + Fixes leak in rtp-payloading unit test. + +2015-07-07 22:23:57 +0900 Hyunjun Ko <zzoonis@gmail.com> + + * gst-libs/gst/audio/gstaudiometa.c: + * gst-libs/gst/video/gstvideometa.c: + * gst-libs/gst/video/video-overlay-composition.c: + video/audio meta: transform_func: return FALSE if not supported or failed + https://bugzilla.gnome.org/show_bug.cgi?id=751778 + +2015-07-07 19:55:44 +0900 Vineeth T M <vineeth.tm@samsung.com> + + * sys/xvimage/xvimagesink.c: + xvimagesink: refactor to use gst_pad_push_event + Right now navigation events are being sent via gst_pad_send_event + after getting the peer pad of the sinkpad. + But the same functionality can be done using gst_pad_push_event + without need of getting peer pad in xvimagesink. + https://bugzilla.gnome.org/show_bug.cgi?id=752059 + +2015-07-07 14:32:25 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/Makefile.am: + * win32/common/libgstvideo.def: + video: Add some more GTypes for enums + +2015-07-02 07:36:12 +0200 Tobias Mueller <muelli@cryptobitch.de> + + * gst-libs/gst/video/video-scaler.c: + GstVideoScaler: Initialised scaling functions to get rid of compiler messages + E.g. + video-scaler.c: In function 'gst_video_scaler_horizontal': + video-scaler.c:1332:3: error: 'func' may be used uninitialized in this function [-Werror=maybe-uninitialized] + func (scale, src, dest, dest_offset, width, n_elems); + ^ + video-scaler.c: In function 'gst_video_scaler_vertical': + video-scaler.c:1373:3: error: 'func' may be used uninitialized in this function [-Werror=maybe-uninitialized] + func (scale, src_lines, dest, dest_offset, width, n_elems); + ^ + GCC's analyses seem to be correct, for the simple fact that if you pass + get_functions a known format, but no hscale or vscale, it'll return + True without having done anything. + Some callers check for the scale values to be not NULL, but then + hscale->resampler.max_taps could return 0. + A different approach to the one presented in this patch is to check + for those max_taps, too, before calling get_functions. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=752051 + +2015-07-07 19:45:43 +0900 Vineeth T M <vineeth.tm@samsung.com> + + * sys/ximage/ximagesink.c: + ximagesink: Post navigation events as message on the bus + post unhandled events to bus, so that + application can utilise the same if needed + https://bugzilla.gnome.org/show_bug.cgi?id=752043 + +2015-07-07 19:35:40 +0900 Vineeth T M <vineeth.tm@samsung.com> + + * sys/ximage/ximagesink.c: + ximagesink: fix navigation event leak + Create event only when pad is created + and send the event to pad. + https://bugzilla.gnome.org/show_bug.cgi?id=752041 + +2015-07-07 09:31:01 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * sys/xvimage/xvimagesink.c: + xvimagesink: fix pad memory leak + pad is not being freed when xwindow is not created + https://bugzilla.gnome.org/show_bug.cgi?id=752042 + +2015-07-07 08:53:09 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * tools/gst-play.c: + gst-play: fix memory leak + In gst-play, for GST_MESSAGE_ELEMENT bus message, + event is being allocated through + gst_navigation_message_parse_event, but not freed. + https://bugzilla.gnome.org/show_bug.cgi?id=752040 + +2015-07-03 21:48:52 +0200 Stefan Sauer <ensonic@users.sf.net> + + * docs/plugins/gst-plugins-base-plugins-sections.txt: + * sys/ximage/ximage.c: + * sys/ximage/ximagepool.c: + * sys/ximage/ximagepool.h: + * sys/ximage/ximagesink.c: + * sys/ximage/ximagesink.h: + * sys/xvimage/xvcontext.c: + * sys/xvimage/xvimage.c: + * sys/xvimage/xvimagepool.c: + * sys/xvimage/xvimagesink.c: + * sys/xvimage/xvimagesink.h: + x/xv_image_sink: rename for consitency + Insert '_' to match the CamelCase. This is needed so that the plugin docs can + guess the names from the type name. + +2015-07-03 21:35:32 +0200 Stefan Sauer <ensonic@users.sf.net> + + * docs/plugins/gst-plugins-base-plugins-docs.sgml: + docs: update master doc for plugins + +2015-07-06 10:05:53 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/typefind/gsttypefindfunctions.c: + typefind: also check moof to recognize video/quicktime + Helps recognizing fragmented files with the right type + +2015-07-06 15:36:07 +0300 Sebastian Dröge <sebastian@centricular.com> + + * docs/libs/gst-plugins-base-libs-sections.txt: + * win32/common/libgstvideo.def: + docs: Add new symbols to the docs and .def files + +2015-07-06 12:53:15 +0300 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/audio-info.h: + * gst-libs/gst/video/video-info.h: + {audio,video}info: Add GST_TYPE_{AUDIO,VIDEO}_INFO macros + +2015-07-06 11:36:58 +0200 Marcin Kolny <marcin.kolny@flytronic.pl> + + * gst-libs/gst/video/video-info.c: + * gst-libs/gst/video/video-info.h: + video-info: implement GstVideoInfo as boxed type + GstVideoInfo usually is created on the stack, but boxed type can be useful + for bindings. + https://bugzilla.gnome.org/show_bug.cgi?id=752011 + +2015-07-02 20:50:00 +0200 Stian Selnes <stian@pexip.com> + + * gst-libs/gst/rtp/gstrtcpbuffer.c: + * tests/check/libs/rtp.c: + rtcpbuffer: Fix validation of packets with padding + The padding (if any) is included in the length of the last packet, see + RFC 3550. + Section 6.4.1: + padding (P): 1 bit + If the padding bit is set, this individual RTCP packet contains + some additional padding octets at the end which are not part of + the control information but are included in the length field. The + last octet of the padding is a count of how many padding octets + should be ignored, including itself (it will be a multiple of + four). + Section A.2: + * The padding bit (P) should be zero for the first packet of a + compound RTCP packet because padding should only be applied, if it + is needed, to the last packet. + * The length fields of the individual RTCP packets must add up to + the overall length of the compound RTCP packet as received. + https://bugzilla.gnome.org/show_bug.cgi?id=751883 + +2015-07-01 17:09:35 +0200 Stian Selnes <stian@pexip.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: Fix setting default pixel-aspect-ratio + It's needed to check if pixel-aspect-ratio exists before fixating. + It does not exist if input caps is not set yet and allowed caps + does not contain pixel-aspect-ratio (e.g. when using GST_VIDEO_CAPS_MAKE) + https://bugzilla.gnome.org/show_bug.cgi?id=751932 + +2015-07-03 21:58:04 +0200 Stefan Sauer <ensonic@users.sf.net> + + * common: + Automatic update of common submodule + From f74b2df to 9aed1d7 + +2015-07-03 21:16:27 +0200 Stefan Sauer <ensonic@users.sf.net> + + * docs/plugins/gst-plugins-base-plugins-sections.txt: + * ext/cdparanoia/gstcdparanoiasrc.h: + * gst/adder/gstadder.h: + * gst/tcp/gstmultisocketsink.h: + docs: order and canonicalize the -sections.txt file + Have all sections in alphabetical order. Also make the macro order consistent. + This is a preparation for generating the file. Remove GET_CLASS macro for + some elements, since it is not used and the header is not installed. + +2015-07-03 21:09:29 +0200 Stefan Sauer <ensonic@users.sf.net> + + * ext/cdparanoia/gstcdparanoiasrc.h: + cdparanoiasrc: remove unused defines + +2015-07-03 21:08:03 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst/videoscale/gstvideoscale.c: + * gst/videoscale/gstvideoscale.h: + videoscale: fix debug categories + Use a local category for the default category and fix the import for the + performance category. + +2015-07-02 10:47:45 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * ext/pango/gstbasetextoverlay.c: + basetextoverlay: Fix bug with unused upstream_has_meta + The intention was to skip the allocation query if upstream has decided + to use the overlay meta feature in the caps. We can safely assume that + upstream have done that query already before making this decision. This + is an optimization since doing allocation queries is relatively + expensive. + CID #1308943 + +2015-07-02 10:27:39 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * ext/pango/gstbasetextoverlay.c: + Revert "basetextoverlay: remove dead code" + This reverts commit e863e5f8a98ceec0ec0bd24274bbae8795e0ab75. + +2015-07-02 14:52:47 +0100 Luis de Bethencourt <luis.bg@samsung.com> + + * ext/pango/gstbasetextoverlay.c: + basetextoverlay: remove dead code + upstream_has_meta is set to FALSE and never changed. The two checks for if + upstream_has_meta will never go to the true branch. Removing the boolean + and the true branches of these checks. + CID #1308943 + +2015-07-02 13:15:58 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: Don't try to get buffers from an empty adapter + +2015-07-01 10:58:07 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/video/gstvideodecoder.c: + * gst-libs/gst/video/gstvideoencoder.c: + {audio,video}{en,de}oder: Also copy POOL metas and make sure to copy over metas when creating subbuffers + POOL meta just means that this specific instance of the meta is related to a + pool, a copy should be made when reasonable and the flag should just not be + set in the copy. + +2015-06-29 18:00:17 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + audiodecoder: Add transform_meta() vfunc with default implementation + The default implementation copies all metadata without tags, and metadata + with only the audio tag. Same behaviour as in GstAudioFilter. + https://bugzilla.gnome.org/show_bug.cgi?id=742385 + +2015-06-29 17:38:38 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/audio/gstaudioencoder.h: + audioencoder: Add transform_meta() vfunc with default implementation + The default implementation copies all metadata without tags, and metadata + with only the audio tag. Same behaviour as in GstAudioFilter. + https://bugzilla.gnome.org/show_bug.cgi?id=742385 + +2015-06-29 15:58:38 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideodecoder.c: + * gst-libs/gst/video/gstvideodecoder.h: + videodecoder: Add transform_meta() vfunc with default implementation + The default implementation copies all metadata without tags, and metadata + with only the video tag. Same behaviour as in GstVideoFilter. + This currently does not work if the ::parse() vfunc is implemented as all + metas are getting lost inside GstAdapter. + https://bugzilla.gnome.org/show_bug.cgi?id=742385 + +2015-06-29 13:59:25 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideoencoder.c: + * gst-libs/gst/video/gstvideoencoder.h: + videoencoder: Add transform_meta() vfunc with default implementation + The default implementation copies all metadata without tags, and metadata + with only the video tag. Same behaviour as in GstVideoFilter. + https://bugzilla.gnome.org/show_bug.cgi?id=742385 + +2015-06-30 10:37:27 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/rtp/gstrtpbaseaudiopayload.c: + rtpbaseaudiopayload: Don't copy memory if not needed, just append payload to the RTP buffer + +2015-06-30 07:26:00 +0900 danny song <danny.song.ga@gmail.com> + + * gst/playback/gstplaybin2.c: + playbin: remove unnecessary break + https://bugzilla.gnome.org/show_bug.cgi?id=751690 + +2015-06-29 16:16:06 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst-libs/gst/video/video-scaler.c: + videoscaler: remove check for below zero for unsigned value + CLAMP checks both if value is '< 0' and '> max'. Value will never be a negative + number since it is a division of an unsigned integer (i). Removing that check + and only checking if it is bigger than max and setting it appropriately. + CID #1308950 + +2015-06-29 13:06:59 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/audioresample/gstaudioresample.c: + audioresample: Also copy metas if their API has no tags attached to it + This is the default basetransform behaviour, being more strict than that + is not really useful. + +2015-06-29 13:06:49 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/audioconvert/gstaudioconvert.c: + audioconvert: Also copy metas if their API has no tags attached to it + This is the default basetransform behaviour, being more strict than that + is not really useful. + +2015-06-29 13:06:33 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudiofilter.c: + audiofilter: Also copy metas if their API has no tags attached to it + This is the default basetransform behaviour, being more strict than that + is not really useful. + +2015-06-29 13:05:54 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideofilter.c: + videofilter: Also copy metas if their API has no tags attached to it + This is the default basetransform behaviour, being more strict than that + is not really useful. + +2015-06-25 00:04:11 +0200 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + Back to development + === release 1.5.2 === -2015-06-24 Sebastian Dröge <slomo@coaxion.net> +2015-06-24 23:24:01 +0200 Sebastian Dröge <sebastian@centricular.com> + * ChangeLog: + * NEWS: + * RELEASE: * configure.ac: - releasing 1.5.2 + * docs/plugins/gst-plugins-base-plugins.args: + * 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-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: + * win32/common/video-enumtypes.c: + * win32/common/video-enumtypes.h: + Release 1.5.2 2015-06-24 22:49:29 +0200 Sebastian Dröge <sebastian@centricular.com> @@ -1,2 +1,2 @@ -This is GStreamer Base Plugins 1.5.2 +This is GStreamer Base Plugins 1.5.90 @@ -1,17 +1,16 @@ -Release notes for GStreamer Base Plugins 1.5.2 +Release notes for GStreamer Base Plugins 1.5.90 -The GStreamer team is pleased to announce the second release of the unstable -1.5 release series. The 1.5 release series is adding new features on top of +The GStreamer team is pleased to announce the first release candidate for the +stable 1.6 release series. The 1.6 release series is adding new features on top of the 1.0, 1.2 and 1.4 series and is part of the API and ABI-stable 1.x release -series of the GStreamer multimedia framework. The unstable 1.5 release series -will lead to the stable 1.6 release series in the next weeks, and newly added -API can still change until that point. +series of the GStreamer multimedia framework. The final 1.6.0 release is planned +in the next few days unless any major bugs are found. -Binaries for Android, iOS, Mac OS X and Windows will be provided separately -during the unstable 1.5 release series. +Binaries for Android, iOS, Mac OS X and Windows will be provided separately by +the GStreamer project. @@ -61,23 +60,38 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg) Bugs fixed in this release - * 708362 : audiobasesink: new slave-method for custom clock slaving algorithms - * 748908 : playsink: cannot enable text flag while playing - * 749039 : x11: X and XV imagesink don't need to cache the pools - * 749243 : textoverlay: gltestsrc ! textoverlay ! fakesink does not work - * 749764 : videoscaler: invalid memory access when downscaling in some cases - * 749823 : basetextoverlay: add properties for toggling drawing text shadow and outline - * 750013 : playbin: seeking halts playback with gst-play-1.0 gapless - * 750455 : xvimagesink, ximagesink: set WM_CLASS for the window - * 750585 : gst-play: sort directory entries - * 750691 : playsink: fix the channel of color balance element - * 750785 : playbin: assrender is not used anymore when available - * 750802 : typefind: UTF-8 MSS manifest parsing support - * 750823 : discoverer test racy - * 751000 : oggdemux: crash with validate.http.media_check.vorbis_theora_1_ogg - * 751118 : playbin: current-suburi does not return correct status when an invalid SUBURI is passed - * 751144 : audioringbuffer: Fix alaw/mulaw channel positions - * 751213 : tools: gst-play: fix seeking issue + * 728636 : textoverlay: cuts off right edge italicised text + * 737427 : appsink: Can't influence allocation query to satisfy user needs + * 742385 : video/audio encoders/decoders: need API to determine when to copy over GstMetas and when to drop them + * 746010 : oggdemux: doesn't go into pull mode even when queue2 ring-buffer is enabled + * 748635 : videorate: caps negotiation regression + * 750235 : [API] rtpbasedepayload: add process_rtp_packet() vfunc + * 751147 : appsink: pull_preroll returns wrong segment in the sample + * 751690 : playbin : remove unnecessary break + * 751883 : rtcpbuffer: Fix validation of packets with padding + * 751932 : GstVideoDecoder: Fix setting default pixel-aspect-ratio + * 752011 : video: Add boxed type for GstVideoInfo structure + * 752040 : gst-play: fix memory leak + * 752041 : ximagesink: fix navigation event leak + * 752042 : xvimagesink: fix pad memory leak + * 752043 : ximagesink: Post navigation events as message on the bus + * 752051 : GstVideoScaler: Initialised scaling functions to get rid of compiler messages + * 752059 : xvimagesink: refactor to use gst_pad_push_event + * 752111 : rtpbasedepayload: Fix minor leak of segment event + * 752156 : playsink: Require streamvolume interface for sink volumes for standardized behaviour + * 752228 : payloader/depayload: Wrong segment handling + * 752359 : playback-test: fix memory leak + * 752397 : playback-test: fix tag list leak + * 752400 : jsseek: fix tag list leak + * 752436 : typefind: Treat *.umx (Unreal Music Package) as audio/x-mod + * 752454 : jsseek: fix memory leaks + * 753001 : rtp buffer: NULL GstBuffer pointer being passed within gst_rtp_buffer_set_extension_data + * 753078 : oggdemux: Segfault on state-change intensive test + * 753088 : decodebin: fix deadend_details string leak + * 753360 : videodecoder: pushes gap before segment + * 753483 : videorate: add support for bayer formats + * 753701 : discoverer: Few trivial fixes in handling error cases + * 611157 : video: API to signal stereoscopic and multiview video ==== Download ==== @@ -114,28 +128,29 @@ subscribe to the gstreamer-devel list. Contributors to this release - * Arun Raghavan - * Brijesh Singh - * Carlos Rafael Giani + * Brian Peters * Edward Hervey - * Guillaume Desmottes + * George Kiagiadakis + * Hyunjun Ko * Jan Schmidt - * Lazar Claudiu + * Joan Pau Beltran + * Lubosz Sarnecki * Luis de Bethencourt - * Lyon Wang - * Matej Knopp + * Marcin Kolny + * Myoungsun Lee * Nicolas Dufresne - * Philippe Normand + * Olivier Crête + * Rico Tzschichholz * Sebastian Dröge - * Song Bing - * Sreerenj Balachandran * Stefan Sauer - * Thibault Saunier + * Stian Selnes + * Thiago Santos * Tim-Philipp Müller + * Tobias Mueller + * Ville Skyttä * Vineeth T M * Vineeth TM * Vivia Nikolaidou - * Víctor Manuel Jáquez Leal * Wim Taymans - * Xavier Claessens + * danny song
\ No newline at end of file diff --git a/configure.ac b/configure.ac index e03e395a7..7e3201149 100644 --- a/configure.ac +++ b/configure.ac @@ -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],[1.5.2.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base]) +AC_INIT([GStreamer Base Plug-ins],[1.5.90],[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, 502, 0, 502) +AS_LIBTOOL(GST, 590, 0, 590) dnl *** required versions of GStreamer stuff *** -GST_REQ=1.5.2.1 +GST_REQ=1.5.90 dnl *** autotools stuff **** diff --git a/docs/plugins/gst-plugins-base-plugins.args b/docs/plugins/gst-plugins-base-plugins.args index 0f5f4a789..34795828f 100644 --- a/docs/plugins/gst-plugins-base-plugins.args +++ b/docs/plugins/gst-plugins-base-plugins.args @@ -2529,6 +2529,16 @@ </ARG> <ARG> +<NAME>GstSocketSrc::caps</NAME> +<TYPE>GstCaps*</TYPE> +<RANGE></RANGE> +<FLAGS>rw</FLAGS> +<NICK>Caps</NICK> +<BLURB>The caps of the source pad.</BLURB> +<DEFAULT></DEFAULT> +</ARG> + +<ARG> <NAME>GstTimeOverlay::time-mode</NAME> <TYPE>GstTimeOverlayTimeLine</TYPE> <RANGE></RANGE> diff --git a/docs/plugins/inspect/plugin-adder.xml b/docs/plugins/inspect/plugin-adder.xml index 3c5e20316..138651c71 100644 --- a/docs/plugins/inspect/plugin-adder.xml +++ b/docs/plugins/inspect/plugin-adder.xml @@ -3,7 +3,7 @@ <description>Adds multiple streams</description> <filename>../../gst/adder/.libs/libgstadder.so</filename> <basename>libgstadder.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-alsa.xml b/docs/plugins/inspect/plugin-alsa.xml index c82f27918..8cc4753cd 100644 --- a/docs/plugins/inspect/plugin-alsa.xml +++ b/docs/plugins/inspect/plugin-alsa.xml @@ -3,7 +3,7 @@ <description>ALSA plugin library</description> <filename>../../ext/alsa/.libs/libgstalsa.so</filename> <basename>libgstalsa.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-app.xml b/docs/plugins/inspect/plugin-app.xml index 9c1b7bc87..0ecb74801 100644 --- a/docs/plugins/inspect/plugin-app.xml +++ b/docs/plugins/inspect/plugin-app.xml @@ -3,7 +3,7 @@ <description>Elements used to communicate with applications</description> <filename>../../gst/app/.libs/libgstapp.so</filename> <basename>libgstapp.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-audioconvert.xml b/docs/plugins/inspect/plugin-audioconvert.xml index ff51de896..d78add14b 100644 --- a/docs/plugins/inspect/plugin-audioconvert.xml +++ b/docs/plugins/inspect/plugin-audioconvert.xml @@ -3,7 +3,7 @@ <description>Convert audio to different formats</description> <filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename> <basename>libgstaudioconvert.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-audiorate.xml b/docs/plugins/inspect/plugin-audiorate.xml index 0364c0947..bdfe33cdf 100644 --- a/docs/plugins/inspect/plugin-audiorate.xml +++ b/docs/plugins/inspect/plugin-audiorate.xml @@ -3,7 +3,7 @@ <description>Adjusts audio frames</description> <filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename> <basename>libgstaudiorate.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-audioresample.xml b/docs/plugins/inspect/plugin-audioresample.xml index eac44e997..8b611dcfe 100644 --- a/docs/plugins/inspect/plugin-audioresample.xml +++ b/docs/plugins/inspect/plugin-audioresample.xml @@ -3,7 +3,7 @@ <description>Resamples audio</description> <filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename> <basename>libgstaudioresample.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-audiotestsrc.xml b/docs/plugins/inspect/plugin-audiotestsrc.xml index 7c0b8a81f..d6a80b777 100644 --- a/docs/plugins/inspect/plugin-audiotestsrc.xml +++ b/docs/plugins/inspect/plugin-audiotestsrc.xml @@ -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>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-cdparanoia.xml b/docs/plugins/inspect/plugin-cdparanoia.xml index 76f76d233..39e6bfd3c 100644 --- a/docs/plugins/inspect/plugin-cdparanoia.xml +++ b/docs/plugins/inspect/plugin-cdparanoia.xml @@ -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>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-encoding.xml b/docs/plugins/inspect/plugin-encoding.xml index 12a9225e4..9f3cedf35 100644 --- a/docs/plugins/inspect/plugin-encoding.xml +++ b/docs/plugins/inspect/plugin-encoding.xml @@ -3,7 +3,7 @@ <description>various encoding-related elements</description> <filename>../../gst/encoding/.libs/libgstencodebin.so</filename> <basename>libgstencodebin.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-gio.xml b/docs/plugins/inspect/plugin-gio.xml index 939de97d7..93e75539f 100644 --- a/docs/plugins/inspect/plugin-gio.xml +++ b/docs/plugins/inspect/plugin-gio.xml @@ -3,7 +3,7 @@ <description>GIO elements</description> <filename>../../gst/gio/.libs/libgstgio.so</filename> <basename>libgstgio.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-libvisual.xml b/docs/plugins/inspect/plugin-libvisual.xml index de826466d..95d9ac231 100644 --- a/docs/plugins/inspect/plugin-libvisual.xml +++ b/docs/plugins/inspect/plugin-libvisual.xml @@ -3,7 +3,7 @@ <description>libvisual visualization plugins</description> <filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename> <basename>libgstlibvisual.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-ogg.xml b/docs/plugins/inspect/plugin-ogg.xml index 34c32bfdc..b10802863 100644 --- a/docs/plugins/inspect/plugin-ogg.xml +++ b/docs/plugins/inspect/plugin-ogg.xml @@ -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>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> @@ -110,7 +110,7 @@ <longname>OGM audio stream parser</longname> <class>Codec/Decoder/Audio</class> <description>parse an OGM audio header and stream</description> - <author>GStreamer maintainers <gstreamer-devel@lists.sourceforge.net></author> + <author>GStreamer maintainers <gstreamer-devel@lists.freedesktop.org></author> <pads> <caps> <name>sink</name> @@ -131,7 +131,7 @@ <longname>OGM text stream parser</longname> <class>Codec/Decoder/Subtitle</class> <description>parse an OGM text header and stream</description> - <author>GStreamer maintainers <gstreamer-devel@lists.sourceforge.net></author> + <author>GStreamer maintainers <gstreamer-devel@lists.freedesktop.org></author> <pads> <caps> <name>sink</name> @@ -152,7 +152,7 @@ <longname>OGM video stream parser</longname> <class>Codec/Decoder/Video</class> <description>parse an OGM video header and stream</description> - <author>GStreamer maintainers <gstreamer-devel@lists.sourceforge.net></author> + <author>GStreamer maintainers <gstreamer-devel@lists.freedesktop.org></author> <pads> <caps> <name>sink</name> diff --git a/docs/plugins/inspect/plugin-pango.xml b/docs/plugins/inspect/plugin-pango.xml index 790f9b1d9..57aae1932 100644 --- a/docs/plugins/inspect/plugin-pango.xml +++ b/docs/plugins/inspect/plugin-pango.xml @@ -3,7 +3,7 @@ <description>Pango-based text rendering and overlay</description> <filename>../../ext/pango/.libs/libgstpango.so</filename> <basename>libgstpango.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> @@ -62,7 +62,7 @@ <longname>Text renderer</longname> <class>Filter/Editor/Video</class> <description>Renders a text string to an image bitmap</description> - <author>David Schleef <ds@schleef.org>, GStreamer maintainers <gstreamer-devel@lists.sourceforge.net></author> + <author>David Schleef <ds@schleef.org>, GStreamer maintainers <gstreamer-devel@lists.freedesktop.org></author> <pads> <caps> <name>sink</name> diff --git a/docs/plugins/inspect/plugin-playback.xml b/docs/plugins/inspect/plugin-playback.xml index 0a0246e37..054d13862 100644 --- a/docs/plugins/inspect/plugin-playback.xml +++ b/docs/plugins/inspect/plugin-playback.xml @@ -3,7 +3,7 @@ <description>various playback elements</description> <filename>../../gst/playback/.libs/libgstplayback.so</filename> <basename>libgstplayback.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-subparse.xml b/docs/plugins/inspect/plugin-subparse.xml index 83d980af2..c6317b47a 100644 --- a/docs/plugins/inspect/plugin-subparse.xml +++ b/docs/plugins/inspect/plugin-subparse.xml @@ -3,7 +3,7 @@ <description>Subtitle parsing</description> <filename>../../gst/subparse/.libs/libgstsubparse.so</filename> <basename>libgstsubparse.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> @@ -35,7 +35,7 @@ <longname>Subtitle parser</longname> <class>Codec/Parser/Subtitle</class> <description>Parses subtitle (.sub) files into text streams</description> - <author>Gustavo J. A. M. Carneiro <gjc@inescporto.pt>, GStreamer maintainers <gstreamer-devel@lists.sourceforge.net></author> + <author>Gustavo J. A. M. Carneiro <gjc@inescporto.pt>, GStreamer maintainers <gstreamer-devel@lists.freedesktop.org></author> <pads> <caps> <name>sink</name> diff --git a/docs/plugins/inspect/plugin-tcp.xml b/docs/plugins/inspect/plugin-tcp.xml index edff6ea63..b1f5cc3d4 100644 --- a/docs/plugins/inspect/plugin-tcp.xml +++ b/docs/plugins/inspect/plugin-tcp.xml @@ -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>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-theora.xml b/docs/plugins/inspect/plugin-theora.xml index d1940670a..8648dac07 100644 --- a/docs/plugins/inspect/plugin-theora.xml +++ b/docs/plugins/inspect/plugin-theora.xml @@ -3,7 +3,7 @@ <description>Theora plugin library</description> <filename>../../ext/theora/.libs/libgsttheora.so</filename> <basename>libgsttheora.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-typefindfunctions.xml b/docs/plugins/inspect/plugin-typefindfunctions.xml index f514c090c..75e05e0d7 100644 --- a/docs/plugins/inspect/plugin-typefindfunctions.xml +++ b/docs/plugins/inspect/plugin-typefindfunctions.xml @@ -3,7 +3,7 @@ <description>default typefind functions</description> <filename>../../gst/typefind/.libs/libgsttypefindfunctions.so</filename> <basename>libgsttypefindfunctions.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-videoconvert.xml b/docs/plugins/inspect/plugin-videoconvert.xml index b84534f2b..3d3addab9 100644 --- a/docs/plugins/inspect/plugin-videoconvert.xml +++ b/docs/plugins/inspect/plugin-videoconvert.xml @@ -3,7 +3,7 @@ <description>Colorspace conversion</description> <filename>../../gst/videoconvert/.libs/libgstvideoconvert.so</filename> <basename>libgstvideoconvert.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> @@ -14,7 +14,7 @@ <longname>Colorspace converter</longname> <class>Filter/Converter/Video</class> <description>Converts video from one colorspace to another</description> - <author>GStreamer maintainers <gstreamer-devel@lists.sourceforge.net></author> + <author>GStreamer maintainers <gstreamer-devel@lists.freedesktop.org></author> <pads> <caps> <name>sink</name> diff --git a/docs/plugins/inspect/plugin-videorate.xml b/docs/plugins/inspect/plugin-videorate.xml index 99d5859f7..71fa47994 100644 --- a/docs/plugins/inspect/plugin-videorate.xml +++ b/docs/plugins/inspect/plugin-videorate.xml @@ -3,7 +3,7 @@ <description>Adjusts video frames</description> <filename>../../gst/videorate/.libs/libgstvideorate.so</filename> <basename>libgstvideorate.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> @@ -20,13 +20,13 @@ <name>sink</name> <direction>sink</direction> <presence>always</presence> - <details>video/x-raw(ANY); image/jpeg(ANY); image/png(ANY)</details> + <details>video/x-raw(ANY); video/x-bayer(ANY); image/jpeg(ANY); image/png(ANY)</details> </caps> <caps> <name>src</name> <direction>source</direction> <presence>always</presence> - <details>video/x-raw(ANY); image/jpeg(ANY); image/png(ANY)</details> + <details>video/x-raw(ANY); video/x-bayer(ANY); image/jpeg(ANY); image/png(ANY)</details> </caps> </pads> </element> diff --git a/docs/plugins/inspect/plugin-videoscale.xml b/docs/plugins/inspect/plugin-videoscale.xml index 664f0c322..5cf1d1f31 100644 --- a/docs/plugins/inspect/plugin-videoscale.xml +++ b/docs/plugins/inspect/plugin-videoscale.xml @@ -3,7 +3,7 @@ <description>Resizes video</description> <filename>../../gst/videoscale/.libs/libgstvideoscale.so</filename> <basename>libgstvideoscale.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-videotestsrc.xml b/docs/plugins/inspect/plugin-videotestsrc.xml index 6fec0c2c2..fc2b5c120 100644 --- a/docs/plugins/inspect/plugin-videotestsrc.xml +++ b/docs/plugins/inspect/plugin-videotestsrc.xml @@ -3,7 +3,7 @@ <description>Creates a test video stream</description> <filename>../../gst/videotestsrc/.libs/libgstvideotestsrc.so</filename> <basename>libgstvideotestsrc.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-volume.xml b/docs/plugins/inspect/plugin-volume.xml index bf342f65b..25071e1c8 100644 --- a/docs/plugins/inspect/plugin-volume.xml +++ b/docs/plugins/inspect/plugin-volume.xml @@ -3,7 +3,7 @@ <description>plugin for controlling audio volume</description> <filename>../../gst/volume/.libs/libgstvolume.so</filename> <basename>libgstvolume.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-vorbis.xml b/docs/plugins/inspect/plugin-vorbis.xml index c4a302093..4c2bc57c4 100644 --- a/docs/plugins/inspect/plugin-vorbis.xml +++ b/docs/plugins/inspect/plugin-vorbis.xml @@ -3,7 +3,7 @@ <description>Vorbis plugin library</description> <filename>../../ext/vorbis/.libs/libgstvorbis.so</filename> <basename>libgstvorbis.so</basename> - <version>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> @@ -41,7 +41,7 @@ <name>sink</name> <direction>sink</direction> <presence>always</presence> - <details>audio/x-raw, format=(string)F32LE, layout=(string)interleaved, rate=(int)[ 1, 200000 ], channels=(int)[ 1, 255 ]</details> + <details>audio/x-raw, format=(string)F32LE, layout=(string)interleaved, rate=(int)[ 1, 200000 ], channels=(int)1; audio/x-raw, format=(string)F32LE, layout=(string)interleaved, rate=(int)[ 1, 200000 ], channels=(int)2, channel-mask=(bitmask)0x0000000000000003; audio/x-raw, format=(string)F32LE, layout=(string)interleaved, rate=(int)[ 1, 200000 ], channels=(int)3, channel-mask=(bitmask)0x0000000000000007; audio/x-raw, format=(string)F32LE, layout=(string)interleaved, rate=(int)[ 1, 200000 ], channels=(int)4, channel-mask=(bitmask)0x0000000000000033; audio/x-raw, format=(string)F32LE, layout=(string)interleaved, rate=(int)[ 1, 200000 ], channels=(int)5, channel-mask=(bitmask)0x0000000000000037; audio/x-raw, format=(string)F32LE, layout=(string)interleaved, rate=(int)[ 1, 200000 ], channels=(int)6, channel-mask=(bitmask)0x000000000000003f; audio/x-raw, format=(string)F32LE, layout=(string)interleaved, rate=(int)[ 1, 200000 ], channels=(int)7, channel-mask=(bitmask)0x0000000000000d0f; audio/x-raw, format=(string)F32LE, layout=(string)interleaved, rate=(int)[ 1, 200000 ], channels=(int)8, channel-mask=(bitmask)0x0000000000000c3f; audio/x-raw, format=(string)F32LE, layout=(string)interleaved, rate=(int)[ 1, 200000 ], channels=(int)[ 9, 255 ], channel-mask=(bitmask)0x0000000000000000</details> </caps> <caps> <name>src</name> diff --git a/docs/plugins/inspect/plugin-ximagesink.xml b/docs/plugins/inspect/plugin-ximagesink.xml index f75c7a666..202171467 100644 --- a/docs/plugins/inspect/plugin-ximagesink.xml +++ b/docs/plugins/inspect/plugin-ximagesink.xml @@ -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>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/docs/plugins/inspect/plugin-xvimagesink.xml b/docs/plugins/inspect/plugin-xvimagesink.xml index cff3be940..118265e6b 100644 --- a/docs/plugins/inspect/plugin-xvimagesink.xml +++ b/docs/plugins/inspect/plugin-xvimagesink.xml @@ -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>1.5.2</version> + <version>1.5.90</version> <license>LGPL</license> <source>gst-plugins-base</source> <package>GStreamer Base Plug-ins source release</package> diff --git a/gst-plugins-base.doap b/gst-plugins-base.doap index 15a299465..014267a53 100644 --- a/gst-plugins-base.doap +++ b/gst-plugins-base.doap @@ -36,6 +36,16 @@ A wide range of video and audio decoders, encoders, and filters are included. <release> <Version> + <revision>1.5.90</revision> + <branch>1.5</branch> + <name></name> + <created>2015-08-19</created> + <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.5.90.tar.xz" /> + </Version> + </release> + + <release> + <Version> <revision>1.5.2</revision> <branch>1.5</branch> <name></name> diff --git a/win32/common/_stdint.h b/win32/common/_stdint.h index 37f816490..61abb1215 100644 --- a/win32/common/_stdint.h +++ b/win32/common/_stdint.h @@ -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 1.5.2" -/* generated using gnu compiler Debian clang version 3.7.0-svn239806-1+b1 (trunk) (based on LLVM 3.7.0) */ +#define _GENERATED_STDINT_H "gst-plugins-base 1.5.90" +/* generated using gnu compiler gcc-5 (Debian 5.2.1-15) 5.2.1 20150808 */ #define _STDINT_HAVE_STDINT_H 1 #include <stdint.h> #endif diff --git a/win32/common/config.h b/win32/common/config.h index 458bcc397..0e2946d57 100644 --- a/win32/common/config.h +++ b/win32/common/config.h @@ -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 "2015-06-24" +#define GST_PACKAGE_RELEASE_DATETIME "2015-08-19" /* Define if static plugins should be built */ #undef GST_PLUGIN_BUILD_STATIC @@ -337,7 +337,7 @@ #define PACKAGE_NAME "GStreamer Base Plug-ins" /* Define to the full name and version of this package. */ -#define PACKAGE_STRING "GStreamer Base Plug-ins 1.5.2" +#define PACKAGE_STRING "GStreamer Base Plug-ins 1.5.90" /* Define to the one symbol short name of this package. */ #define PACKAGE_TARNAME "gst-plugins-base" @@ -346,7 +346,7 @@ #undef PACKAGE_URL /* Define to the version of this package. */ -#define PACKAGE_VERSION "1.5.2" +#define PACKAGE_VERSION "1.5.90" /* directory where plugins are located */ #ifdef _DEBUG @@ -380,7 +380,7 @@ #undef USE_TREMOLO /* Version number of package */ -#define VERSION "1.5.2" +#define VERSION "1.5.90" /* Define WORDS_BIGENDIAN to 1 if your processor stores words with the most significant byte first (like Motorola and SPARC, unlike Intel). */ diff --git a/win32/common/video-enumtypes.c b/win32/common/video-enumtypes.c index 9263cb456..e82b178aa 100644 --- a/win32/common/video-enumtypes.c +++ b/win32/common/video-enumtypes.c @@ -14,6 +14,8 @@ #include "video-tile.h" #include "video-converter.h" #include "video-resampler.h" +#include "video-frame.h" +#include "video-scaler.h" /* enumerations from "video-format.h" */ GType @@ -803,3 +805,93 @@ gst_video_resampler_flags_get_type (void) } return g_define_type_id__volatile; } + +/* enumerations from "video-frame.h" */ +GType +gst_video_frame_flags_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GFlagsValue values[] = { + {GST_VIDEO_FRAME_FLAG_NONE, "GST_VIDEO_FRAME_FLAG_NONE", "none"}, + {GST_VIDEO_FRAME_FLAG_INTERLACED, "GST_VIDEO_FRAME_FLAG_INTERLACED", + "interlaced"}, + {GST_VIDEO_FRAME_FLAG_TFF, "GST_VIDEO_FRAME_FLAG_TFF", "tff"}, + {GST_VIDEO_FRAME_FLAG_RFF, "GST_VIDEO_FRAME_FLAG_RFF", "rff"}, + {GST_VIDEO_FRAME_FLAG_ONEFIELD, "GST_VIDEO_FRAME_FLAG_ONEFIELD", + "onefield"}, + {GST_VIDEO_FRAME_FLAG_MULTIPLE_VIEW, "GST_VIDEO_FRAME_FLAG_MULTIPLE_VIEW", + "multiple-view"}, + {GST_VIDEO_FRAME_FLAG_FIRST_IN_BUNDLE, + "GST_VIDEO_FRAME_FLAG_FIRST_IN_BUNDLE", "first-in-bundle"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_flags_register_static ("GstVideoFrameFlags", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + +GType +gst_video_buffer_flags_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GFlagsValue values[] = { + {GST_VIDEO_BUFFER_FLAG_INTERLACED, "GST_VIDEO_BUFFER_FLAG_INTERLACED", + "interlaced"}, + {GST_VIDEO_BUFFER_FLAG_TFF, "GST_VIDEO_BUFFER_FLAG_TFF", "tff"}, + {GST_VIDEO_BUFFER_FLAG_RFF, "GST_VIDEO_BUFFER_FLAG_RFF", "rff"}, + {GST_VIDEO_BUFFER_FLAG_ONEFIELD, "GST_VIDEO_BUFFER_FLAG_ONEFIELD", + "onefield"}, + {GST_VIDEO_BUFFER_FLAG_MULTIPLE_VIEW, + "GST_VIDEO_BUFFER_FLAG_MULTIPLE_VIEW", "multiple-view"}, + {GST_VIDEO_BUFFER_FLAG_FIRST_IN_BUNDLE, + "GST_VIDEO_BUFFER_FLAG_FIRST_IN_BUNDLE", "first-in-bundle"}, + {GST_VIDEO_BUFFER_FLAG_LAST, "GST_VIDEO_BUFFER_FLAG_LAST", "last"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_flags_register_static ("GstVideoBufferFlags", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + +GType +gst_video_frame_map_flags_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GFlagsValue values[] = { + {GST_VIDEO_FRAME_MAP_FLAG_NO_REF, "GST_VIDEO_FRAME_MAP_FLAG_NO_REF", + "no-ref"}, + {GST_VIDEO_FRAME_MAP_FLAG_LAST, "GST_VIDEO_FRAME_MAP_FLAG_LAST", "last"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_flags_register_static ("GstVideoFrameMapFlags", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + +/* enumerations from "video-scaler.h" */ +GType +gst_video_scaler_flags_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GFlagsValue values[] = { + {GST_VIDEO_SCALER_FLAG_NONE, "GST_VIDEO_SCALER_FLAG_NONE", "none"}, + {GST_VIDEO_SCALER_FLAG_INTERLACED, "GST_VIDEO_SCALER_FLAG_INTERLACED", + "interlaced"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_flags_register_static ("GstVideoScalerFlags", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} diff --git a/win32/common/video-enumtypes.h b/win32/common/video-enumtypes.h index 986fe1bf7..090e7ee05 100644 --- a/win32/common/video-enumtypes.h +++ b/win32/common/video-enumtypes.h @@ -89,6 +89,18 @@ GType gst_video_resampler_method_get_type (void); #define GST_TYPE_VIDEO_RESAMPLER_METHOD (gst_video_resampler_method_get_type()) GType gst_video_resampler_flags_get_type (void); #define GST_TYPE_VIDEO_RESAMPLER_FLAGS (gst_video_resampler_flags_get_type()) + +/* enumerations from "video-frame.h" */ +GType gst_video_frame_flags_get_type (void); +#define GST_TYPE_VIDEO_FRAME_FLAGS (gst_video_frame_flags_get_type()) +GType gst_video_buffer_flags_get_type (void); +#define GST_TYPE_VIDEO_BUFFER_FLAGS (gst_video_buffer_flags_get_type()) +GType gst_video_frame_map_flags_get_type (void); +#define GST_TYPE_VIDEO_FRAME_MAP_FLAGS (gst_video_frame_map_flags_get_type()) + +/* enumerations from "video-scaler.h" */ +GType gst_video_scaler_flags_get_type (void); +#define GST_TYPE_VIDEO_SCALER_FLAGS (gst_video_scaler_flags_get_type()) G_END_DECLS #endif /* __GST_VIDEO_ENUM_TYPES_H__ */ |