diff options
| author | Sebastian Dröge <sebastian@centricular.com> | 2014-05-03 17:50:10 +0200 |
|---|---|---|
| committer | Sebastian Dröge <sebastian@centricular.com> | 2014-05-03 17:50:10 +0200 |
| commit | 68f5350c664b52ea2e77b1852df5795bae20c758 (patch) | |
| tree | dc1955ccee5245d2a231a4b9a4e7e2b1dd46df7d | |
| parent | 876e28b9468a9f6fc9574fa9395e85599cf9e15a (diff) | |
Release 1.3.11.3.1
47 files changed, 3869 insertions, 229 deletions
@@ -1,9 +1,3442 @@ +=== release 1.3.1 === + +2014-05-03 Sebastian Dröge <slomo@coaxion.net> + + * configure.ac: + releasing 1.3.1 + +2014-05-03 17:22:10 +0200 Sebastian Dröge <sebastian@centricular.com> + + * 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/hr.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: + po: Update translations + +2014-05-02 19:09:59 -0400 Olivier Crête <olivier.crete@collabora.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + * tests/check/libs/rtpbasepayload.c: + rtpbasepayload: Implement reconfigure event & renegotiation without subclass + Implement the reconfigure event, also do correct downstream caps negotiation + if the subclass doesn't implementy set_caps. + https://bugzilla.gnome.org/show_bug.cgi?id=725361 + +2014-05-02 19:09:44 -0400 Olivier Crête <olivier.crete@collabora.com> + + * tests/check/libs/rtpbasepayload.c: + tests/check/libs/rtpbasepayload.c: Run gst-indent + https://bugzilla.gnome.org/show_bug.cgi?id=725361 + +2014-05-03 10:14:51 +0200 Sebastian Dröge <sebastian@centricular.com> + + * common: + Automatic update of common submodule + From bcb1518 to 211fa5f + +2014-05-02 18:30:16 -0400 Olivier Crête <olivier.crete@collabora.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: Save the PT after fixating + +2014-05-02 19:36:34 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/rtsp/gstrtspdefs.c: + * gst-libs/gst/rtsp/gstrtspdefs.h: + rtspdefs: remove outdated comments + +2014-05-02 15:09:35 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst-libs/gst/rtp/gstrtpbuffer.c: + rtpbuffer: avoid underflow in size calculation + +2014-05-01 19:31:09 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: do not parse caps for not using it + Saving some cpu + +2014-01-03 11:06:22 +0100 John Bassett <john.bassett@pexip.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: restrict initial random sequence number to be <= 32767 + In order to prevent SRTP roll over counter issues the initial sequence + number is restricted to <= 32767. This is recommended by RFC 4568 section 6.4. + +2014-05-01 15:11:04 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/sdp/gstsdpmessage.c: + sdp: Add some more gobject-introspection annotations for bindings + https://bugzilla.gnome.org/show_bug.cgi?id=729123 + +2014-05-01 13:15:57 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplaybin2.c: + playbin: Don't block on non-serialized events + https://bugzilla.gnome.org/show_bug.cgi?id=729321 + +2014-05-01 13:08:24 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplaysink.c: + playsink: Don't block on non-serialized events + https://bugzilla.gnome.org/show_bug.cgi?id=729321 + +2014-05-01 13:06:53 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: Don't block on non-serialized events + https://bugzilla.gnome.org/show_bug.cgi?id=729321 + +2014-05-01 13:05:05 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: Don't block on non-serialized events + https://bugzilla.gnome.org/show_bug.cgi?id=729321 + +2014-04-30 11:06:27 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst-libs/gst/rtp/gstrtcpbuffer.c: + rtcpbuffer: check claimed data size against available size + Coverity 1208773 + +2014-04-23 08:06:36 +0200 Göran Jönsson <goranjn@axis.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: Empty queue when flush. + Empty the watchs queue when calling + gst_rtsp_watch_set_flushing with flushing variabel is TRUE. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728772 + +2014-03-16 16:09:36 +0100 Ognyan Tonchev <otonchev@gmail.com> + + * tests/check/libs/rtspconnection.c: + rtspconnection: Add more tests + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728907 + +2014-04-29 10:15:47 -0400 Luis de Bethencourt <luis@debethencourt.com> + + * gst/videotestsrc/videotestsrc.c: + videotestsrc: fix undefined behaviour of left-shift + With a small type for the color values being left-shifted, the result is + undefined and it could potentially overflow. + https://bugzilla.gnome.org/show_bug.cgi?id=729195 + +2014-04-29 10:59:02 +0100 Tim-Philipp Müller <tim@centricular.com> + + * win32/common/libgstrtsp.def: + * win32/common/libgstsdp.def: + win32: fix export files again + Revert unintended parts of d8a0927930a87a2eb60d4c98cb3fea8aed911b27 + +2014-04-29 11:39:18 +0200 Christian Fredrik Kalager Schaller <uraeus@linuxrising.org> + + * gst-plugins-base.spec.in: + * win32/common/libgstrtsp.def: + * win32/common/libgstsdp.def: + Add mikey.h file + +2014-04-29 09:58:21 +0200 Haakon Sporsheim <haakon@pexip.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: Make caps writable before fixating + https://bugzilla.gnome.org/show_bug.cgi?id=729114 + +2014-04-29 09:54:18 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/sdp/gstsdpmessage.c: + sdpmessage: Add array length annotation to gst_sdp_message_parse_buffer + https://bugzilla.gnome.org/show_bug.cgi?id=729123 + +2014-04-29 08:46:02 +0200 Stian Selnes <stian@pexip.com> + + * gst-libs/gst/rtp/gstrtpbuffer.c: + rtpbuffer: fix memory leak when gst_rtp_buffer_map fails + Make sure rtp->data[3] is set before jumping to error path. + https://bugzilla.gnome.org/show_bug.cgi?id=729117 + +2014-04-28 18:47:06 +0530 Ravi Kiran K N <ravi.kiran@samsung.com> + + * tools/gst-play.c: + gst-play: add option to supply media files from playlist file + https://bugzilla.gnome.org/show_bug.cgi?id=728845 + +2014-04-27 00:49:01 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/gio/gstgiobasesink.c: + giobasesink: we mustn't change the format of a query response + Not even in the DEFAULT case. That's bad 0.10 behaviour, no caller + is ever going to check the format of the response. + +2014-04-27 00:25:16 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/playback/gstplay-enum.c: + playbin: add nick for soft colorbalance play flag to fix gst-inspect + Fix gst-inspect-1.0 playbin criticals when printing the + flags, which was caused by a missing nick name for one + of the flags. + +2014-04-26 23:26:09 +0100 Tim-Philipp Müller <tim@centricular.com> + + * ext/alsa/gstalsasink.c: + * ext/alsa/gstalsasrc.c: + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggmux.c: + * ext/theora/gsttheoradec.c: + * ext/theora/gsttheoraenc.c: + * ext/theora/gsttheoraparse.c: + * ext/vorbis/gstvorbisdec.c: + * ext/vorbis/gstvorbisenc.c: + * ext/vorbis/gstvorbisparse.c: + * gst-libs/gst/app/gstappsink.c: + * gst-libs/gst/app/gstappsrc.c: + * gst-libs/gst/audio/gstaudiobasesink.c: + * gst-libs/gst/audio/gstaudiobasesrc.c: + * gst-libs/gst/audio/gstaudioclock.c: + * gst-libs/gst/audio/gstaudiofilter.c: + * gst-libs/gst/audio/gstaudioringbuffer.c: + * gst-libs/gst/audio/gstaudiosink.c: + * gst-libs/gst/audio/gstaudiosrc.c: + * gst-libs/gst/rtp/gstrtcpbuffer.c: + * gst-libs/gst/rtp/gstrtpbuffer.c: + * gst-libs/gst/rtp/gstrtphdrext.c: + * gst-libs/gst/rtp/gstrtppayloads.c: + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtspdefs.c: + * gst-libs/gst/rtsp/gstrtspextension.c: + * gst-libs/gst/rtsp/gstrtspmessage.c: + * gst-libs/gst/rtsp/gstrtsprange.c: + * gst-libs/gst/rtsp/gstrtsptransport.c: + * gst-libs/gst/rtsp/gstrtspurl.c: + * gst-libs/gst/sdp/gstmikey.c: + * gst-libs/gst/sdp/gstsdpmessage.c: + * gst/adder/gstadder.c: + * gst/audioconvert/gstaudioconvert.c: + * gst/playback/gstplaybin2.c: + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultihandlesink.c: + * gst/tcp/gstmultioutputsink.c: + * gst/tcp/gstmultisocketsink.c: + * gst/videorate/gstvideorate.c: + * gst/videoscale/gstvideoscale.c: + docs: remove outdated and pointless 'Last reviewed' lines from docs + They are very confusing for people, and more often than not + also just not very accurate. Seeing 'last reviewed: 2005' in + your docs is not very confidence-inspiring. Let's just remove + those comments. + +2014-04-25 17:32:59 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/gio/gstgiobasesink.c: + giobasesink: Implement handling of the SEEKING query + +2014-04-25 11:30:37 +0200 Edward Hervey <bilboed@bilboed.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: Plug caps leaks + We were returning in various places without unreffing the caps, and + we were also leaking (overwriting) the caps we got from _get_current_caps() + Spotted by Haakon Sporsheim in #gstreamer + +2014-04-22 18:28:10 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/audioresample/resample.c: + audioresample: Don't left-shift into the sign bit, instead use unsigned integers + +2014-04-22 00:21:01 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/tag/gstexiftag.c: + tag: exif: avoid adding empty strings + Fixes assertion with some jpeg files + +2014-04-21 15:35:32 +0200 Wim Taymans <wtaymans@redhat.com> + + * tools/gst-play.c: + play: Improve pipeline states + First set the pipeline to the PAUSED state to check if we are dealing + with a live pipeline or not. Then move to the desired state. + If we don't do this, it is possible that we receive a BUFFERING message + before we know that the pipeline is live and we would set the pipeline + to PAUSED and deadlock. + +2014-04-21 15:33:10 +0200 Wim Taymans <wtaymans@redhat.com> + + * tools/gst-play.c: + play: Update buffering state for live pipelines + Update the buffering variable, even for live pipelines so that we don't + print \n for each buffering message. + +2014-04-16 19:53:14 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/video-frame.c: + videoframe: Initialise GstVideoFrame to zeroes if mapping fails + This should allow for more meaningful errors. Dereferencing NULL + is more useful information than dereferencing a random address + happened to be on the stack. + +2014-04-16 11:43:40 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst-libs/gst/tag/gstexiftag.c: + exiftag: catch buffer mapping failure + Might be what caused: + Coverity 1139734 + +2014-04-15 19:17:06 +0200 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/audioresample.c: + audioresample: Fix memory leaks in test + +2014-04-15 19:16:44 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/audioresample/gstaudioresample.c: + * gst/audioresample/resample.c: + audioresample: Fix up indention + +2014-04-15 19:16:18 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/audioresample/resample_sse.h: + audioresample: Fix out of bounds memory accesses + +2014-04-15 13:57:08 +0200 Sebastian Dröge <sebastian@centricular.com> + + * ext/pango/gstbasetextoverlay.c: + pango: Make static caps actually static to fix a memory leak + +2014-04-15 13:54:45 +0200 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/videotestsrc.c: + videotestsrc: Fix memory leak in test + +2014-04-15 13:48:46 +0200 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/encodebin.c: + encodebin: Fix memory leak in test + +2014-04-15 13:48:17 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/pbutils/encoding-profile.c: + encoding-profile: Free preset name in finalize + +2014-04-15 13:39:39 +0200 Sebastian Dröge <sebastian@centricular.com> + + * ext/ogg/gstoggmux.c: + oggmux: Clear Ogg streams before initing them + They might've been inited before, in which case we leak + memory when initing them again without clearing. + +2014-04-15 13:03:34 +0200 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/audioconvert.c: + audioconvert: Fix leaks in unit test + +2014-04-15 11:55:22 +0200 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/libs/videodecoder.c: + * tests/check/libs/videoencoder.c: + videoencoder/decoder: Fix memory leaks in the tests + +2014-04-15 11:53:43 +0200 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/libs/audiodecoder.c: + audiodecoder: Actually allocate enough memory for 64 bits, not just 32 bits + Also fix a memory leak. + +2014-04-15 11:43:41 +0200 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/libs/audioencoder.c: + audioencoder: Fix memory leaks in unit test + +2014-04-15 10:29:12 +0200 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/libs/rtp.c: + rtp: Fix GBytes memory leak in test + +2014-04-12 07:10:36 +0200 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + rtpbasedepay: add stats property + Add a stats property that holds a structure with all the current + values of the depayloader. + See https://bugzilla.gnome.org/show_bug.cgi?id=646577 + +2014-04-12 06:43:24 +0200 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: update docs + +2014-04-12 06:27:36 +0200 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: add current timestamp and seqnum offset to stats + Expose the current timestamp and seqnum offset in the stats + See https://bugzilla.gnome.org/show_bug.cgi?id=646577 + +2014-04-11 10:24:10 +0200 Josep Torra <n770galaxy@gmail.com> + + * ext/pango/gsttextrender.c: + * ext/pango/gsttextrender.h: + textrender: push segment event after caps event + Fixes warning "Sticky event misordering, got 'segment' before 'caps'". + +2014-04-10 16:08:29 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/ogg/gstoggstream.c: + oggstream: use G_GUINT64_CONSTANT instead of ll suffix + Thanks slomo for pointing out it's not standard. + +2014-04-10 15:55:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * sys/xvimage/xvcontext.c: + xvimage: remove dead code + matching_attr can not be NULL here, we've tested that away a few + lines beforehand. + Coverity 1139655 + +2014-04-10 15:51:05 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst/videotestsrc/gstvideotestsrc.c: + videotestsrc: bail out on unsupported caps + This avoids using uninitialized data (and properly rejects caps). + Coverity 1139898 + +2014-04-10 15:16:03 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst/typefind/gsttypefindfunctions.c: + typefind: remove pointless checks for data being NULL + It was already checked in an early out, and as it's only + incremented for at most the size of the passed buffer, it + can only become NULL in an address wraparound. + While there, don't cast away const on a pointer. + Coverity 1139845 + +2014-04-10 13:34:58 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst/playback/gstdecodebin2.c: + decodebin: consider "no demuxer" case to not have dynamic pads + This fixes a possible NULL dereference. + Coverity 1195146 + +2014-04-10 13:28:30 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst/encoding/gstencodebin.c: + encodebin: guard against gst_pad_get_peer returning NULL + If it does, the pad may be leaked if it's a request pad, though. + Coverity 1139799 + +2014-04-10 13:26:42 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst/encoding/gstencodebin.c: + encodebin: guard against pathological NULL dereference + Coverity 1139798 + +2014-04-10 12:32:24 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst/audioresample/resample.c: + audioresample: reject 0 denominator when creating resampler + Coverity 1195140, 1195139, 1195138 + +2014-04-10 12:14:48 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst-libs/gst/video/video-overlay-composition.c: + video-overlay-composition: guard against NULL pointer dereference on error + If gst_video_overlay_rectangle_apply_global_alpha is called with + a rectangle with unsuitable alpha, expanding the alpha plane will + fail, and thus lead to dereferencing a NULL src pointer. It's not + certain this will happen in practice, as the function is static + and callers might ensure suitable alpha before calling, but there + is no apparent explicit such check. + Add prologue asserts for proper alpha to explicitely prevent this. + Coverity 1139707 + +2014-04-10 12:10:47 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst-libs/gst/video/gstvideometa.c: + videometa: fix texture_type memcpy size + Coverity 1139589, 1139588 + +2014-04-10 11:19:26 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst-libs/gst/sdp/gstsdpmessage.c: + sdpmessage: fix multi statement macros + Wasn't playing nice with an if statement below. + Coverity 1139767 + +2014-04-10 11:14:25 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst-libs/gst/audio/gstaudiocdsrc.c: + audiocdsrc: guard aginst overflow + An audio CD may contain about a tenth of the samples 32 bit can + represent, so it doesn't seem likely this will be hit in practice. + Coverity 1139805 + +2014-04-10 12:30:50 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/pbutils/descriptions.c: + pbutils: descriptions: default to systemstream=false for partial video/mpeg caps + Assume systemstream=false for video/mpeg caps where that field + is missing. + +2014-04-10 10:57:53 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst-libs/gst/audio/gstaudiobasesink.c: + audiobasesink: avoid possible sample count overflow + At 48 kHz, 2<<31 samples is reached before 13 hours so it + sounds plausible this would be hit. + Coverity 1139800, 1139801 + +2014-04-10 10:45:21 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/theora/gsttheoraenc.c: + theoraenc: fix comparison to unset timestamp + Also rejects negative timestamps that aren't GST_CLOCK_TIME_NONE. + Coverity 1139797 + +2014-04-10 10:33:46 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/ogg/gstoggstream.c: + oggstream: fix a few left shifts operations on 32 bits cast to 64 bits + This should not cause any actual bug since Theora and Daala have + a maximum shift of 31, and a packet duration of 2^31 seems very + implausible. But it fixes: + Coverity 1139804, 1139803, 1139802 + +2014-04-10 10:29:34 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/ogg/gstoggstream.c: + oggstream: remove NULL test after dereference + And add NULLness asserts at top of function. The only call + to this passes local variable pointers, so non NULL. + Coverity 206375 + +2014-04-10 10:25:46 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/ogg/gstoggmux.c: + oggmux: test for failure to return tag + It should really not happen unless the tag list it corrupt, + but the API returns a failure code so we may as well use it. + Coverity 1139595 + +2014-04-10 10:22:43 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/ogg/gstoggdemux.c: + oggdemux: do not dereference NULL pad in warning message + Coverity 1197695 + +2014-04-10 09:18:05 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/video-event.c: + video-event: Update the running times in the force-keyunit events from the pad offsets + +2014-04-09 16:03:15 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: In adaptive streaming mode, only have a fixed buffer limit for the non-buffering multiqueue + +2014-04-08 15:43:50 +0200 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/sdp/gstsdpmessage.c: + sdp: guard against address parse errors. + +2014-03-25 17:11:34 +0100 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com> + + * gst/adder/gstadder.c: + adder: rework the logic to check if eos has to be sent. + Checking the size available was incorrect, and the infos + for per-pad EOS are available. + Same logic as audiomixer. + fixes: https://bugzilla.gnome.org/show_bug.cgi?id=727025 + +2014-04-08 12:46:21 +0200 Josep Torra <n770galaxy@gmail.com> + + * gst-libs/gst/audio/gstaudioringbuffer.c: + audioringbuffer: parse channels field from compressed audio caps + Also parse channels as an optional field in the caps for compressed + audio formats. + +2014-04-06 22:26:20 +1000 Jan Schmidt <jan@centricular.com> + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: Consider all caps for overlays, not just the first. + Check all supported caps on the overlay video pad, not just the + first of (possibly) many. + +2014-04-05 13:25:46 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-play-1.0.1: + tools: update gst-play-1.0 man page + +2014-04-02 07:20:43 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: do not deactivate the bufferpool, just unref + Videodecoder does late renegotiation, it will wait for the next + buffer before renegotiating its caps and bufferpool. It might happen + that downstream element switched from passthrough to non-passthrough + and sent a reconfigure upstream (that caused this renegotiation). + This downstream element will ask the video sink below for the bufferpool + with an allocation query and will get the same bufferpool that + videodecoder is holding, too. + When renegotiating, if videodecoder deactivates its bufferpool it + might be deactivating the bufferpool that some element downstream + is using and cause the pipeline to fail. + https://bugzilla.gnome.org/show_bug.cgi?id=727498 + +2014-02-24 11:17:05 -0500 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst-libs/gst/audio/gstaudiobasesink.c: + audiobasesink: clip start samples to match clipped start time + Clock slaving can clip start time to zero, giving us a shorted + duration than we originally got. To keep in sync, we must then + discard the samples falling before that zero timestamp. + This possibly fixes random distortion caused by constant PA + underflows which are never resynced. + +2014-04-04 17:36:04 +0200 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/sdp/gstmikey.c: + * gst-libs/gst/sdp/gstmikey.h: + * tests/check/libs/mikey.c: + * win32/common/libgstsdp.def: + mikey: Fix the KEMAC payload + The KEMAC payload actually needs to have subpayloads and the key should + go into the KEY_DATA subpayload. Add support for subpayloads and + implement the KEY_DATA payload. + Add some pointers to the conversion functions that allow us to add + encryption and decryption later. + +2014-04-04 02:14:50 +1100 Jan Schmidt <jan@centricular.com> + + * gst/playback/gstplaybin2.c: + playbin: Drop reference to any source element in NULL state + Drop the reference instead of waiting for either finalize(), or + for a new source when reused. Everyone else already forgot about + the old source. + +2014-04-01 10:38:23 +0200 Göran Jönsson <goranjn@axis.com> + + * win32/common/libgstrtsp.def: + rtspconnection: Added gst_rtsp_watch_set_flushing to list. + Added gst_rtsp_watch_set_flushing to list in file + libgstrtsp.def + +2014-03-30 18:26:59 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: Always drain the decoder after a discont group in reverse playback mode + +2014-03-30 17:54:11 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: Flush the decoder once per discont group, not once per keyframe + +2014-03-30 17:54:11 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: Handle reverse playback with multiple GOPs per discont group properly + baseparse will reverse each GOP for us already, so the segment events can + be after our keyframe. Make sure to get it and all other relevant sticky + events before starting to decode. + +2014-03-29 10:23:05 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: Log event types of events that are pushed downstream + +2014-03-27 20:15:01 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: In reverse playback mode we need to finish the subclass after passing all frames to it + +2014-03-28 09:32:20 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtspconnection.h: + rtspconnection: add flush method + Add a method to set/unset the flushing state that makes _wait_backlog() + unlock. + See https://bugzilla.gnome.org/show_bug.cgi?id=725898 + +2014-03-27 16:43:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * sys/ximage/ximagesink.c: + ximagesink: only extrapolate alpha mask for 32-bit depth + Instead of passing bogus alpha mask values when there's no alpha. + https://bugzilla.gnome.org/show_bug.cgi?id=727188 + +2014-03-25 11:14:51 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/sdp/gstmikey.c: + mikey: fix return values of g_return_* + +2014-03-25 11:07:34 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/rtsp/gstrtsptransport.c: + rtsptransport: UDP is also default for SAVP and AVPF + +2014-03-20 12:29:33 +0100 Wim Taymans <wtaymans@redhat.com> + + * docs/libs/gst-plugins-base-libs-docs.sgml: + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/sdp/gstmikey.c: + * gst-libs/gst/sdp/gstmikey.h: + docs: add MIKEY docs + +2014-03-15 18:46:52 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/sdp/Makefile.am: + * gst-libs/gst/sdp/gstmikey.c: + * gst-libs/gst/sdp/gstmikey.h: + * tests/check/Makefile.am: + * tests/check/libs/mikey.c: + * win32/common/libgstsdp.def: + mikey: add MIKEY parsing helpers + MIKEY is defined in RFC 3830 and is used to exchange SRTP encryption + parameters between a sender and a receiver in a secure way. + This library implements a subset of the features, enough to implement + RFC 4567, using MIKEY in SDP and RTSP. + +2014-03-16 17:04:44 +0100 Ognyan Tonchev <otonchev@gmail.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: Fix minor memory leaks in error handling + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726642 + +2014-03-16 17:06:02 +0100 Ognyan Tonchev <otonchev@gmail.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: Fix connection_poll() + * Only check for conditions we are interested in. + * Makes no sense to specify G_IO_ERR and G_IO_HUP in condition, they + will always be reported if they are true. + * Do not create timed source if timeout is NULL. + * Correctly wait for sources to be dispatched, context_iteration() is + not guaranteed to always block even if set to do so. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726641 + +2014-03-20 09:18:31 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: add pt and ssrc to stats + +2014-03-16 08:34:30 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/elements/decodebin.c: + * tests/check/elements/decodebin2.c: + tests: decodebin: port old decodebin2 test for parser and decoder linking + They were in the old decodebin2.c tests file and were never ported. + Now we can get rid of decodebin2.c + +2014-03-16 17:00:38 +0100 Arun Raghavan <arun@accosted.net> + + * gst/playback/gstplay-enum.c: + * gst/playback/gstplay-enum.h: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gstplaysink.h: + * tests/examples/playback/playback-test.c: + playback: Add video-/audio-filter properties + This provides an audio-filter and video-filter property to allow + applications to set filter elements/bins. The idea is that these will + e + applied if possible -- for non-raw sinks, the filters will be skipped. + If the application wishes to force the application of the filters, this + can be done by setting the new flag introduced on playsink - + GST_PLAY_FLAG_FORCE_FILTERS. + https://bugzilla.gnome.org/show_bug.cgi?id=679031 + +2014-03-16 18:38:25 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplay-enum.h: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gstplaysink.h: + Revert "playback: Add video-/audio-filter properties" + This reverts commit fb8fdedb4f4649aa33700bbc720131c1678df49f. + +2014-03-15 16:05:22 +0100 Arun Raghavan <arun.raghavan@collabora.co.uk> + + * gst/playback/gstplay-enum.h: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gstplaysink.h: + playback: Add video-/audio-filter properties + This provides an audio-filter and video-filter property to allow + applications to set filter elements/bins. The idea is that these will be + applied if possible -- for non-raw sinks, the filters will be skipped. + If the application wishes to force the application of the filters, this + can be done by setting the new flag introduced on playsink - + GST_PLAY_FLAG_FORCE_FILTERS. + https://bugzilla.gnome.org/show_bug.cgi?id=679031 + +2014-03-15 20:21:32 +0000 Руслан Ижбулатов <lrn1986@gmail.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: Silence a compiler warning + Cast the argument into (const char *) on W32, as winsock2 expects it. + https://bugzilla.gnome.org/show_bug.cgi?id=726433 + +2014-03-15 11:24:23 +0100 Arun Raghavan <arun.raghavan@collabora.co.uk> + + * gst/playback/gstplaysink.c: + playsink: Fix documentation for what the audio chain looks like + https://bugzilla.gnome.org/show_bug.cgi?id=679031 + +2014-03-11 21:58:49 +0000 Tim-Philipp Müller <tim@centricular.com> + + * docs/plugins/gst-plugins-base-plugins.args: + * 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-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: + docs: update plugin docs and remove old properties and signals + Re-generate .args and .signals file from scratch so that + old signals that no longer exist (such as the 'new-decoded-pad' + signal on decodebin) no longer show up in the documentation. + +2014-03-11 22:15:13 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst/adder/gstadder.c: + adder: set a group-id on the stream-start event + Set a default group-id to fix a warning printed by the sink. + +2014-03-11 17:39:54 +0100 Christian Fredrik Kalager Schaller <uraeus@linuxrising.org> + + * gst-plugins-base.spec.in: + Add new header file + +2014-03-06 12:59:08 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggmux.c: + * ext/ogg/gstoggstream.c: + * ext/ogg/gstoggstream.h: + oggmux: implement vp8 granulepos function + Add an extra function to the oggstream map to inform it about + the incoming buffers. This way oggmux can keep a count on the + vp8 invisible frames and calculate the granulepos correctly. + https://bugzilla.gnome.org/show_bug.cgi?id=722682 + +2014-03-05 16:34:42 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * ext/ogg/gstoggmux.c: + * ext/ogg/gstoggstream.c: + * ext/ogg/gstoggstream.h: + oggmux: create vp8 header data if not provided in caps + vp8 stream header shouldn't be assumed to be provided in caps always + as this would repeat the same code in all demuxers/encoders. Instead, + make oggmux generate them if they are not supplied. + https://bugzilla.gnome.org/show_bug.cgi?id=722682 + +2014-03-06 13:55:17 +0100 Göran Jönsson <goranjn@axis.com> + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtspconnection.h: + * win32/common/libgstrtsp.def: + rtspconnection: gst_rtsp_watch_wait_backlog + New method that wait until there is room in backlog queue. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898 + +2014-03-06 13:50:27 +0100 David Svensson Fors <davidsf@axis.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtspconnection.h: + rtspconnection: GstRTSPWatch func for tunnel GET response + Add a callback in GstRTSPWatch where the response to HTTP GET for + tunneled connections can be modified. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725878 + +2014-03-06 15:34:47 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/rtsp/gstrtspdefs.c: + * gst-libs/gst/rtsp/gstrtspdefs.h: + rtspdefs: add RFC 4567 headers and status code + This new Header and status code is used for SRTP + +2014-03-07 17:09:24 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + * gst/playback/gsturidecodebin.c: + decodebin: Buffer up to 5 seconds in multiqueue buffering mode + 2 seconds might be too small for some container formats, e.g. + MPEGTS with some video codec and AAC/ADTS audio with 700ms + long buffers. The video branch of multiqueue can run full while + the audio branch is completely empty, especially because there + are usually more queues downstream on the audio branch. + +2014-03-06 22:37:44 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: Keep the number of buffers after an adaptive streaming demuxer lower + Usually these buffers are multiple seconds large, and having a maximum + of 5 buffers in the multiqueue there can use a lot of memory. Lower + this to 2 for adaptive streaming demuxers. + +2014-03-06 22:28:46 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: Simplify adaptive streaming demuxer code a bit + +2014-03-06 17:49:09 +0000 Adrien Schwartzentruber <adrien.schwartzentruber@gmail.com> + + * ext/pango/gstbasetextoverlay.c: + pango: demote debug WARNING to LOG for variable framerate video input + No need why we need to warn about that, it's perfectly allowed. + https://bugzilla.gnome.org/show_bug.cgi?id=725837 + +2014-01-30 15:41:49 +0000 Matthieu Bouron <matthieu.bouron@collabora.com> + + * tests/check/Makefile.am: + * tests/check/elements/textoverlay.c: + tests: add textoverlay passthrough with composition feature unit tests + https://bugzilla.gnome.org/show_bug.cgi?id=721953 + +2014-01-23 12:20:05 +0000 Matthieu Bouron <matthieu.bouron@collabora.com> + + * ext/pango/gstbasetextoverlay.c: + pango: basetextoverlay: handle video/x-raw(ANY) if downstream supports the GstVideoOverlayCompositionMeta API + https://bugzilla.gnome.org/show_bug.cgi?id=721953 + +2014-01-23 12:19:13 +0000 Matthieu Bouron <matthieu.bouron@collabora.com> + + * gst-libs/gst/video/video-overlay-composition.h: + video-overlay-composition: add GST_CAPS_FEATURE_META_GST_VIDEO_OVERLAY_COMPOSITION + +2014-03-04 16:51:58 +0200 Andres Gomez <agomez@igalia.com> + + * REQUIREMENTS: + * docs/plugins/gst-plugins-base-plugins.args: + * docs/plugins/gst-plugins-base-plugins.signals: + docs: Removing GnomeVFS left bits + gnomevfs was removed time ago but there are still some left bits. + https://bugzilla.gnome.org/show_bug.cgi?id=725658 + +2014-03-05 00:35:30 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/typefind/gsttypefindfunctions.c: + typefindfunctions: lower H.263 typefinder max probability + The typefinder returns LIKELY for as little as one possible + sync and no bad sync (not even taking into account how much + data was looked at for that). It's generally just not fit + for purpose, so should just not return anything like LIKELY + at all ever, even more so since it only recognises one out + of ten H263 files, and likes to mis-detect mp3s as H263. + https://bugzilla.gnome.org/show_bug.cgi?id=700770 + https://bugzilla.gnome.org/show_bug.cgi?id=725644 + +2014-03-02 11:58:58 +0100 Ognyan Tonchev <ognyan@axis.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + * tests/check/libs/rtspconnection.c: + rtspconnection: Call closed() when GET is closed in tunneled mode + This patch adds read source on the write socket in tunneled + mode and we get a callback when client disconnects the GET + channel. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725313 + +2014-03-02 12:58:21 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * gst-libs/gst/video/video-format.c: + videoformat: Remove duplicate/incorrect section + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521 + +2014-03-02 12:54:08 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtsptransport.c: + * gst-libs/gst/rtsp/gstrtspurl.c: + * gst-libs/gst/video/video-format.c: + docs: Add annotations for return values + Rephrase and clarify some return value descriptions + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521 + +2014-03-02 05:06:07 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + docs: Fix argument and annotation typos + * colorbalance: Fix misspelled annotation + * rtsp: Replace incorrectly documented function argument + * sdp: Escape @ character to avoid gtk-doc warning + * video-*: Add missing annotation colon + * videodecoder/video-color: Fix function argument typos + * videoutils: Remove unknown annotation field + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521 + +2014-03-02 05:09:05 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * .gitignore: + .gitignore: Ignore gcov intermediate files + https://bugzilla.gnome.org/show_bug.cgi?id=725479 + +2014-02-28 09:34:31 +0100 Sebastian Dröge <sebastian@centricular.com> + + * common: + Automatic update of common submodule + From fe1672e to bcb1518 + +2014-02-20 20:01:30 +0000 Matthieu Bouron <matthieu.bouron@collabora.com> + + * gst/playback/gstplaybin2.c: + playbin: improve autoplug_query_caps return + Makes autoplug_query_caps return + downstream_caps + intersect_first(filter_caps, element_caps) + https://bugzilla.gnome.org/show_bug.cgi?id=724828 + +2014-02-26 22:11:01 +0100 Stefan Sauer <ensonic@users.sf.net> + + * common: + Automatic update of common submodule + From 1a07da9 to fe1672e + +2014-02-26 11:43:06 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtsp: fix build with older GLib versions + The gio/gnetworking.h header is only available since glib 2.36 + https://bugzilla.gnome.org/show_bug.cgi?id=725206 + +2014-02-26 11:45:24 +0100 Ognyan Tonchev <ognyan@axis.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: Add missing include + https://bugzilla.gnome.org/show_bug.cgi?id=725206 + +2014-02-21 14:01:37 +0000 Matthieu Bouron <matthieu.bouron@collabora.com> + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: improve gst_play_sink_convert_bin_getcaps return + If we have the peer caps and a caps filter, return peer_caps + + intersect_first (filter, converter_caps) instead of + intersect_first (filter, peer_caps + converter_caps) and preservers + downstream caps preference order. + https://bugzilla.gnome.org/show_bug.cgi?id=724893 + +2014-01-31 00:06:18 +0100 Sebastian Rasmussen <sebrn@axis.com> + + * tests/check/Makefile.am: + * tests/check/libs/.gitignore: + * tests/check/libs/rtp-basepayloading.c: + * tests/check/libs/rtpbasedepayload.c: + * tests/check/libs/rtpbasepayload.c: + tests: Refactor RTP basepayloading test into pay/depay parts + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723328 + +2014-01-31 00:19:16 +0100 Sebastian Rasmussen <sebrn@axis.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: Let caps event also configure seqnum-offset + Previously the sequence number kept track of by GstRTPBasePayload would + only be set when going from READY to PAUSED state. This meant that a + downstream element that attempted to configure a basepayloader by + setting seqnum-offset e.g. in its sinkpad's caps template would have + trouble configuring the basepayloader. The reason was that the caps + event which arrives with the desired value for seqnum-offset did not + arrive at the basepayloader until caps negotiation took place, + significantly later than the transition from READY to PAUSED. + The result after this patch is that the default value for the + seqnum-offset property, or later set values for this property, will take + effect when going from READY to PAUSED like before. In addition the an + arriving caps event will also affect the basepayloaders configured + sequence number as the event arrives. + +2014-01-31 00:18:35 +0100 Sebastian Rasmussen <sebrn@axis.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: Fix payload type property boundary value + The payload type field in an RTP packet header is 7 bits wide, hence the + boundary values ought to be 0x00 and 0x7f, not the previously stated + values 0x00 and 0x80. + +2014-01-31 00:06:30 +0100 Sebastian Rasmussen <sebrn@axis.com> + + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + rtpbasedepayload: Fix typos in comments + +2014-02-21 19:28:55 +0000 Tim-Philipp Müller <tim@centricular.com> + + * docs/libs/gst-plugins-base-libs-docs.sgml: + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/video/gstvideopool.c: + docs: add GstVideoPool to docs + +2014-02-21 09:53:09 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: If we have a demuxer without dynamic srcpads, just assume no-more-pads + Otherwise we will wait until the multiqueue after the demuxer will + overrun, which is clearly not needed then. + +2014-02-21 09:43:38 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: Also make sure to not duplicate an element factory after a group + If we are using an adaptive stream demuxer, which outputs a non-container + stream, we are putting another multiqueue after the *parser* following + the adaptive stream demuxer. We do not want to add another instance of + the same parser right after this multiqueue. + +2014-02-20 15:38:48 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: During pre-rolling always use the auto-preroll limits on multiqueues + Even if we're buffering in the multiqueues. + +2014-02-20 15:37:54 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: Pass through the seekability information when setting multiqueue limits + +2014-02-20 15:36:47 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: During exposing of pads don't set the multiqueue limits multiple times to different values + Instead just set them once in the very end to the correct values. + +2014-02-20 15:07:26 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: Only enable multiqueue buffering once we're pre-rolled + Otherwise we will emit buffering messages not just from the last + multiqueue but also from previous multiqueues... confusing the + application with different percentages during pre-rolling. + +2014-02-20 15:02:09 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstdecodebin2.c: + decodebin: Make sure that we always have a second multiqueue for adaptive streaming demuxers + For adaptive streaming demuxer we insert a multiqueue after + this demuxer. This multiqueue will get one fragment per buffer. + Now for the case where we have a container stream inside these + buffers, another demuxer will be plugged and after this second + demuxer there will be a second multiqueue. This second multiqueue + will get smaller buffers and will be the one emitting buffering + messages. + If we don't have a container stream inside the fragment buffers, + we'll insert a multiqueue below right after the next element after + the adaptive streaming demuxer. This is going to be a parser or + decoder, and will output smaller buffers. + +2014-02-19 10:21:16 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gsturidecodebin.c: + uridecodebin: Always use buffering in multiqueue for adaptive streams + +2014-02-19 10:06:13 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gsturidecodebin.c: + uridecodebin: Only add a queue2 for buffering for non-adaptive streaming streams + +2013-02-06 08:46:58 -0300 Thiago Santos <thiago.sousa.santos@collabora.com> + + * gst/playback/gsturidecodebin.c: + uridecodebin: pass on the buffering property for adaptive streams + Adaptive streams should download its data inside the demuxer, so + we want to use multiqueue's buffering messages to control the + pipeline flow and avoid losing sync if download rates are low; + https://bugzilla.gnome.org/show_bug.cgi?id=707636 + +2014-02-21 19:07:59 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/libs/.gitignore: + tests: add new unit tests to .gitignore + +2014-02-19 13:54:17 +0100 Ognyan Tonchev <ognyan@axis.com> + + * tests/check/Makefile.am: + * tests/check/libs/rtspconnection.c: + rtspconnection: New unit test + See https://bugzilla.gnome.org/show_bug.cgi?id=724720 + +2014-02-19 13:53:06 +0100 Ognyan Tonchev <ognyan@axis.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: Remove read child source when POST is disconnected + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724720 + +2014-02-19 16:10:25 -0800 Aleix Conchillo Flaqué <aleix@oblong.com> + + * win32/common/libgstrtsp.def: + defs: update for new rtspconnection symbols + +2014-02-19 01:55:50 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * ext/ogg/gstoggdemux.c: + oggdemux: allow file to go until the end in push mode + When seeking back to original state after duration seeks, let + upstream know that we want the whole file, including the last + byte that wasn't requested on the duration seeks. + https://bugzilla.gnome.org/show_bug.cgi?id=724633 + +2014-02-19 23:54:59 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggdemux.h: + oggdemux: remove unused instance variable event + It is never set to anything + +2014-02-16 17:39:35 -0800 Aleix Conchillo Flaqué <aleix@oblong.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtspconnection.h: + rtspconnection: allow specifying a certificate database + Two new functions have been added, + gst_rtsp_connection_set_tls_database() and + gst_rtsp_connection_get_tls_database(). The certificate database will be + used when a certificate can't be verified with the default database. + https://bugzilla.gnome.org/show_bug.cgi?id=724393 + +2014-02-16 23:55:17 -0800 Aleix Conchillo Flaqué <aleix@oblong.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: get rid of superfluous whitespaces + +2014-02-18 20:48:57 +0100 Stefan Sauer <ensonic@users.sf.net> + + * tests/check/elements/encodebin.c: + encodebin: simplify tests + Also use the profile helper for the ogg profile here. + +2014-02-18 13:08:09 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * gst-libs/gst/video/video-info.c: + video: Fix NV12_64Z32 default offset and size + This was a regression introduced by f52fd7a68, where we started using + the stride to encode the dimensions in tiles. This patch simply updates + offset and size calculation as described in the documentation, + part-mediatype-video-raw.txt. + +2014-02-18 15:02:57 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplaybin2.c: + playbin: Keep inputselector around until we release its pads + Otherwise there's an interesting race condition when we destroy + the inputselector (actually it will be destroyed later when its state + change message gets destroyed) and afterwards release its sinkpad. + This is the code path when the last channel is removed from the + input selector. + Gave this warning sometimes, for chained oggs or whenever else + we change decode groups: + GStreamer-CRITICAL **: Padname '':sink_0 does not belong to element inputselector0 when removing + +2014-02-18 10:42:04 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/audioconvert/gstchannelmix.c: + audioconvert: never do mixing for 1->1 channel conversions + MONO and NONE position are the same, for example, but in + general there isn't much to do here for such a conversion. + Fixes problem in audioconvert, which would end up using + a mixmatrix when converting between different mono format + because it thinks MONO positioning is different from + unpositioned channels, which is not the case in this + special case. The mixmatrix would end up being 0.0 so + audioconvert would convert to silence samples. + https://bugzilla.gnome.org/show_bug.cgi?id=724509 + +2014-02-18 10:32:46 +0000 Rafał Mużyło <galtgendo@o2.pl> + + * gst-libs/gst/audio/audio-info.c: + audio: map channels=1,channel-mask=0 to MONO instead of NONE + Fixes problem in audioconvert, which would end up using + a mixmatrix when converting between different mono format + because it thinks MONO positioning is different from + unpositioned channels, which is not the case in this + special case. The mixmatrix would end up being 0.0 so + audioconvert would convert to silence samples. + https://bugzilla.gnome.org/show_bug.cgi?id=724509 + +2014-02-16 21:24:29 +0100 Stefan Sauer <ensonic@users.sf.net> + + * tests/check/elements/encodebin.c: + encodebin: refactor tests + Add a new test to demo how to get missing plugin message. + Split some tests that unneccesarily munge unrelated checks into one test. + +2014-02-16 15:32:47 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplaysink.c: + playsink: Only remove the complete text chain if the text pad goes away + If the text pads does not go away we just set the overlay to silent, which + allows us to immediately re-enable subs later again. However before this + change we also released the streamsynchronizer text pads, which deadlocked + because there was still dataflow going on. Just do this only if we remove + the complete chain. + https://bugzilla.gnome.org/show_bug.cgi?id=683504 + +2014-02-14 20:16:04 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tools/Makefile.am: + * tools/gst-play.c: + tools: gst-play: add volume control + +2014-02-13 16:03:01 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * ext/ogg/gstoggmux.c: + oggmux: properly flush when seeking at the beginning + Reset all internal status when collect pads forwards a flush-stop + from the pads to be able to start the stream again. + +2014-02-12 17:34:32 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gsturidecodebin.c: + uridecodebin: Don't leak pad references + +2014-02-02 23:59:36 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * tests/check/Makefile.am: + tests: Don't build disabled plugins' check tests + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723492 + +2014-02-11 16:35:45 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplaybin2.c: + playbin: First try to get the pad's current caps, then query caps + The caps query might give us ANY caps while the pad has fixed caps + configured currently. + +2014-02-10 16:33:50 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplaybin2.c: + playbin: Fix memory leak in autoplugging code + We should not leak element factories ideally. + +2014-02-10 16:33:35 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/playbin-complex.c: + playbin: Fix memory leak in unit test + +2014-02-09 23:17:03 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: Remove unused function + +2014-02-09 11:28:48 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudiosrc.h: + audiosrc: Fix typo in docs + We read *from* the audio device, not to it. + +2014-02-08 17:11:54 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/videoscale.c: + videoscale: Fix compiler warning in unit test + error: implicit conversion from enumeration type + 'GstFormat' to different enumeration type 'GstVideoFormat' + +2014-02-08 17:11:04 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/videoconvert.c: + videoconvert: Fix compiler warning in unit test + error: implicit conversion from enumeration type + 'GstFormat' to different enumeration type 'GstVideoFormat' + +2014-02-08 17:07:15 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/examples/playback/playback-test.c: + playback-test: Fix types for comparisons + Storing a 64 bit integer in a 32 bit integer and then checking + for the error cases might not be ideal. + error: comparison of constant -9223372036854775808 with + expression of type 'guint' (aka 'unsigned int') is always true + +2014-02-08 17:02:27 +0100 Sebastian Dröge <sebastian@centricular.com> + + * ext/ogg/gstoggmux.h: + oggmux: Fix typo in header include guard + clang does not like this. + +2014-02-08 17:01:38 +0100 Sebastian Dröge <sebastian@centricular.com> + + * ext/alsa/gstalsaplugin.c: + alsa: Make clang happy with our g_strdup_vprintf() wrapper + +2014-02-07 15:33:34 +0100 Wim Taymans <wtaymans@redhat.com> + + * tests/examples/playback/playback-test.c: + playback-test: allow seeking outside of the range + For download buffer, allow seeking outside of the already downloaded + area. + +2014-02-07 02:09:10 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * ext/pango/gstbasetextoverlay.c: + basetextoverlay: use correct segment for text + video time uses the 'segment' and the text time should use + the 'text_segment'. + If different segments are used for video and text it would + lead to out of sync video/subtitles. + +2014-02-04 14:31:29 +0100 Wim Taymans <wtaymans@redhat.com> + + * tests/check/libs/rtp.c: + check: add some more checks + Add header and payload length check in case of CSRCs. + See https://bugzilla.gnome.org/show_bug.cgi?id=723196 + +2014-02-03 02:35:57 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * tests/examples/seek/jsseek.c: + jsseek: Add missing HAVE_X check + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723507 + +2014-02-04 13:55:49 +0100 Eric Trousset <etrousset@awox.com> + + * gst-libs/gst/tag/gsttagdemux.c: + tagdemux: Forward TIME seeks upstream too, maybe upstream can handle that + https://bugzilla.gnome.org/show_bug.cgi?id=723597 + +2014-01-31 23:27:03 +0100 Stefan Sauer <ensonic@users.sf.net> + + * docs/libs/gst-plugins-base-libs-docs.sgml: + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/audio/audio-channels.c: + * gst-libs/gst/audio/gstaudiometa.c: + docs: doc fixes for audio library + Add sections docs for audiometa. Fix sections docs for audiochannels. Remove old + mixerutil section. + +2014-01-31 13:40:36 +0000 Julien Isorce <julien.isorce@collabora.co.uk> + + * gst/videotestsrc/gstvideotestsrc.c: + videotestsrc: ensure having caps when setting the buffer pool config + It happens if downstream does not propose a buffer pool. + GST_DEBUG=2 gst-launch-1.0 videotestsrc ! fakesink + https://bugzilla.gnome.org/show_bug.cgi?id=723271 + +2014-01-30 21:18:04 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tools/gst-play.c: + gst-play: Support non-ASCII tags + By calling setlocale() to get us multi-byte/UTF-8 support. + https://bugzilla.gnome.org/show_bug.cgi?id=723164 + +2014-01-28 14:28:27 +0100 Bastien Nocera <hadess@hadess.net> + + * tools/gst-discoverer.c: + gst-discoverer: Support non-ASCII tags + By calling setlocale() to get us multi-byte/UTF-8 support. + https://bugzilla.gnome.org/show_bug.cgi?id=723164 + +2014-01-30 10:43:48 +0100 Edward Hervey <bilboed@bilboed.com> + + * common: + Automatic update of common submodule + From d48bed3 to 1a07da9 + +2014-01-29 13:58:07 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst/encoding/gststreamsplitter.c: + streamsplitter: push pending events before eos + Push any pending events downstream before pushing eos + +2014-01-29 12:33:21 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/Makefile.am: + * tests/check/libs/.gitignore: + * tests/check/libs/audioencoder.c: + tests: audioencoder: add tests analogous to the videoencoder ones + +2014-01-29 12:32:16 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: push pending events and tags before EOS + if there are tags or events pending and an EOS is received, push those + events and tags before the EOS. + +2014-01-28 15:25:05 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/libs/videoencoder.c: + tests: videoencoder: check that tags are pushed before eos + Check that if a new tag event is received right before eos it + is pushed before the eos + +2014-01-28 15:30:35 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/video/gstvideoencoder.c: + videoencoder: push tags and events before eos + if any tags or events are pending, push them before pushing eos + +2014-01-28 15:06:39 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/Makefile.am: + * tests/check/libs/.gitignore: + * tests/check/libs/videoencoder.c: + tests: videoencoder: basic videoencoder base class test + Adds a single test for video encoding + +2013-11-26 01:13:45 +0100 Sebastian Rasmussen <sebrn@axis.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: Do cosmetic changes to rtptime calculations + * Change running time type to guint64 + * Use GST_CLOCK_TIME_NONE() to check for invalid timestamps + * Name variables so ns-based and hz-based timestamps are evident + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383 + +2014-01-28 00:40:38 +0100 Sebastian Rasmussen <sebrn@axis.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: Expose running-time of payloaded stream + https://bugzilla.gnome.org/show_bug.cgi?id=719415 + +2014-01-22 17:47:02 +0100 Sebastian Rasmussen <sebrn@axis.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: Improve documentation for perfect-rtptime + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383 + +2014-01-16 16:58:43 +0100 Sebastian Rasmussen <sebrn@axis.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: Fix typos in documentation for properties + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383 + +2014-01-28 00:19:07 +1100 Alessandro Decina <alessandro.d@gmail.com> + + * gst/playback/gstdecodebin2.c: + * gst/playback/gsturidecodebin.c: + decodebin: make it possible to register multiple handlers for autoplug-select + Change the way autoplug-select is accumulated so that it's possible to have + multiple handlers. The handlers keep getting called as long as they keep + returning GST_AUTOPLUG_SELECT_TRY. + One practical example of when this is needed is when hooking into playbin's + uridecodebin, which is perhaps not very elegant but the only way to influence + which streams playbin autoplugs/exposes. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723096 + +2014-01-16 21:49:59 +0100 Sebastian Rasmussen <sebrn@axis.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + * tests/check/libs/rtp-basepayloading.c: + rtpbasepayload: Add statistics property + This property allows for an atomically retrieved set of properties that + can e.g. be used to generate RTP-Info headers. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719415 + +2013-07-26 15:44:28 +0200 Sjoerd Simons <sjoerd.simons@collabora.co.uk> + + * gst/playback/gsturidecodebin.c: + uridecodebin: Drop hardcoded list of media suitable for download buffering + Discussion on IRC indicated that the main reason for this list was to + prevent demuxers that can trigger a lot of seeking from using + progressive buffering using queue2 (which due to being seekable triggers + that behaviour). + However given that upstream can indicate seeks are possible but should + be avoided via a scheduling query, this extra whitelisting shouldn't be + necessary for well-behaved demuxers. + https://bugzilla.gnome.org/show_bug.cgi?id=704933 + +2014-01-24 12:19:43 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/videoconvert/gstvideoconvert.c: + videoconvert: tweak the scoring algorithm + Make a little table of conversions and manually score them. Use this + info to define better weights for the scoring algorithm. + give separate scores for doing changes and the impact of the change, + This allows us to avoid conversion when we can but still allow fairly + lossless changes. + The old code did not penalize GRAY conversions, PAL conversions were + punished too low and depth conversions too high. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722656 + +2014-01-23 10:45:00 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/video-chroma.c: + video-chroma: don't crash on NULL resamplers + Make dummy resamplers for all cases and only execute the horizontal + resampler instead of crashing. + See https://bugzilla.gnome.org/show_bug.cgi?id=722742 + +2014-01-21 11:21:56 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + audiobasesink: make _get_time more threadsafe + We call the _get_time function from the provided clock and we don't lock + the sink object for performance reasons. Make sure we only read and + check variables once so that they don't change while we are executing + the code. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720661 + +2014-01-20 16:11:04 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/audioresample/resample.c: + audioresample: It's HAVE_EMMINTRIN_H, not HAVE_XMMINTRIN_H for SSE2 + +2014-01-20 15:44:09 +0100 Antoine Jacoutot <ajacoutot@gnome.org> + + * gst/audioresample/resample.c: + audioresample: Fix build on x86 if emmintrin.h is available but can't be used + On i386, EMMINTRIN is defined but not usable without SSE so check for + __SSE__ and __SSE2__ as well. + https://bugzilla.gnome.org/show_bug.cgi?id=670690 + +2014-01-20 10:30:36 +0100 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + configure: Initialize Qt variables + +2014-01-20 09:46:15 +0100 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + * tests/examples/overlay/Makefile.am: + * tests/examples/overlay/qt-videooverlay.cpp: + examples: Port Qt examples to Qt5 + +2014-01-18 19:22:12 +0100 Nicola Murino <nicola.murino@gmail.com> + + * gst-libs/gst/riff/riff-media.c: + riff: Fix G726 caps creation + https://bugzilla.gnome.org/show_bug.cgi?id=720995 + +2014-01-18 00:18:51 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: minor docs fix + Can use a custom main context as well if needed. + +2014-01-18 13:54:22 +0100 Sebastian Dröge <sebastian@centricular.com> + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/video/gstvideodecoder.c: + * gst-libs/gst/video/gstvideodecoder.h: + * win32/common/libgstvideo.def: + videodecoder: Add API to get the currently pending frame size for parsing + https://bugzilla.gnome.org/show_bug.cgi?id=719890 + +2014-01-18 21:20:51 +0900 Wonchul Lee <chul0812@gmail.com> + + * gst/playback/gstplaybin2.c: + playbin: Remove unnecessary assignment + Remove duplicated assignment + https://bugzilla.gnome.org/show_bug.cgi?id=722491 + +2014-01-18 13:31:06 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplaybin2.c: + playbin: Insert decoders without GstAVElement information between the other decoders + Otherwise they would be preferred over all decoders independent + of their ranks. + https://bugzilla.gnome.org/show_bug.cgi?id=722316 + +2014-01-18 13:12:16 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplaybin2.c: + playbin: Only put parsers and sinks first, not all non-decoders + https://bugzilla.gnome.org/show_bug.cgi?id=722316 + +2014-01-17 11:08:32 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/libs/videodecoder.c: + tests: videodecoder: plug a few leaks + Remove leaks of caps and events references + +2014-01-17 10:17:29 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: plug leak when frames are released on subclass stop + They end up stored in the 'pending_events' list and should be + freed after calling stop + +2014-01-17 15:10:42 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tools/gst-play.c: + gst-play: Handle CLOCK_LOST message + It is necessary for playbin gapless playback when switching + between audio-only and video-only files for example. + +2014-01-16 16:32:34 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/encoding/gststreamsplitter.c: + streamsplitter: handle ACCEPT_CAPS query correctly + We can accept a caps when one of the downstream peers can accept the + caps. This is not the same as checking a subset of the getcaps + result because parsers might accept broader caps than what their getcaps + function returns (See https://bugzilla.gnome.org/show_bug.cgi?id=677401). + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722330 + +2014-01-14 13:02:28 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/libs/audiodecoder.c: + tests: audiodecoder: add another test for negotiation with gap event + Check that even if the subclass doesn't call set_output_format, the base + class should use upstream provided caps to fill the output caps that is + pushed before the gap event is forwarded, otherwise it ends again fixating + the rate and channels to 1. + https://bugzilla.gnome.org/show_bug.cgi?id=722144 + +2014-01-14 13:05:54 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: copy rate and channels from input before fixating output caps + For default caps generation when handling gap events that are sent + before any buffer, try to use caps that are closer to what upstream + provided to avoid fixating rate or channels to 1 as default. + So there are the steps: + 1) Try to set rate, channels and channel-mask from upstream if provided + 2) Fixate the rate and channels to the default rate and channels from + audio lib + 3) Fixate the caps just to be sure everything is fixed + 4) If no channel-mask was provided and channels > 2, use a default + channel-mask (taken from audioconvert code) + https://bugzilla.gnome.org/show_bug.cgi?id=722144 + +2014-01-14 23:07:34 +0100 Holger Kaelberer <hk@getslash.de> + + * sys/xvimage/xvimagesink.c: + xvimagesink: don't recreate xvcontext + A xvcontext can be created early in gst_xvimagesink_set_window_handle(). + In this case don't recreate, i.e. overwrite it in gst_xvimagesink_open(). + Otherwise XEvents won't be handled in the xevent listener thread. + Fixes a regression when setting the window handle on the sink in + the very beginning before changing its state. + https://bugzilla.gnome.org/show_bug.cgi?id=715138 + +2014-01-14 12:05:46 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/ogg/gstoggdemux.c: + oggdemux: fix broken seeking reading the whole file + A change in gst_ogg_demux_do_seek caused oggdemux to wait for + a page for each of the streams, including a skeleton stream if + one was present. Since Skeleton only has header pages, that + was never going to end well. + Also, the code was skipping CMML streams when looking for pages, + so would also have broken on CMML streams. + Thus, we change the code to disregard Skeleton streams, as well + as discontinuous streams (such as CMML and Kate). While it may + be desirable to consider Kate streams too (in order to avoid + losing a subtitle starting near the seek point), this may be + a performance drag when seeking where no subtitles are. Maybe + one could add a "give up" threshold for such discontinuous + streams, so we'd get any page if there is one, but do not end + up reading preposterous amounts of data otherwise. + In any case, it is important that the code that determines + the amount of streams to look pages for remains consistent with + the "early out" conditions of the code that actually parses + the incoming pages, lest we never decrease the pending counter + to zero. + This fixes seeking on a file with a skeleton track reading all + the file on each seek. + https://bugzilla.gnome.org/show_bug.cgi?id=719615 + +2014-01-13 15:14:14 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggdemux.h: + oggdemux: use an adaptive chunksize for performance reasons + Ogg data is read chunk by chunk, and the chunk size used was + originally taken from libvorbisfile. However, this value leads + to poor performance when used on an Ogg file with large pages + (Ogg pages can be close to 64 KB). + We can't just use a larger chunk size, since this will decrease + performance on small page streams, so we use an adaptive scheme + where the chunk size is twice the largest page size we've seen + so far in the stream. For "typical" Ogg/Vorbis, this gives us + almost the same chunk size (a bit lower), and this lets us get + better performance on streams with large pages. + +2014-01-13 20:47:02 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: avoid parsing caps event if it is not used + Saves some cpu + +2014-01-13 20:44:23 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: make sure caps is set before forwarding gap event + Before trying to generate a default fixated caps when handling a gap + event, make sure that the same strategy that is used when handling + a buffer has been attempted. Otherwise audiodecoder will ignore + upstream caps settings such as rate and channels and will likely + end with a caps with channels=1 and rate=1. + https://bugzilla.gnome.org/show_bug.cgi?id=722144 + +2014-01-13 19:40:49 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/libs/audiodecoder.c: + tests: audiodecoder: check that negotiation works buffers and gaps + Adds 2 tests to verify that output caps are the expected value, reusing + input structure values for both buffers and gaps + https://bugzilla.gnome.org/show_bug.cgi?id=722144 + +2014-01-13 16:33:11 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/Makefile.am: + * tests/check/libs/.gitignore: + * tests/check/libs/audiodecoder.c: + tests: audiodecoder: add basic playback test for audio decoder + Simple test that just check that audio decoding works as expected + https://bugzilla.gnome.org/show_bug.cgi?id=722144 + +2014-01-14 13:17:26 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/videooverlay.c: + videoverlay: Don't mention gconf elements and add a sentence about playbin/playsink + playbin/playsink now implement the video overlay interface + +2014-01-13 16:28:23 +0000 Tim-Philipp Müller <tim@centricular.com> + + * win32/common/libgstvideo.def: + win32: add new API to .def file + +2014-01-13 16:29:00 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: only copy chroma_site when known + Only overwrite the chroma-site if we have a valid value in the reference + format. + +2014-01-13 16:20:55 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/videoconvert/gstvideoconvertorc.orc: + * gst/videoconvert/videoconvert.c: + videoconvert: don't interpolate chroma in I420 -> RGB + Don't try to interpolate the chroma samples, the used algorithm only + works for horizontal cositing. Let's switch to a faster and safer + version until we handle chroma siting correctly in the fastpaths. + +2014-01-13 12:16:01 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/gstvideoutils.c: + videoutils: add some debug + +2014-01-08 19:43:01 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * docs/libs/gst-plugins-base-libs-sections.txt: + doc: Add new sections introduce for tile format + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2014-01-08 19:42:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * gst-libs/gst/video/Makefile.am: + video: Generate types for tile enumeration + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2014-01-08 19:41:56 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * docs/design/part-mediatype-video-raw.txt: + * gst-libs/gst/video/video-format.c: + * gst-libs/gst/video/video-format.h: + * gst-libs/gst/video/video-frame.c: + * gst-libs/gst/video/video-info.c: + * gst-libs/gst/video/video-tile.h: + video: Don't use extra plane and componenent for tile format + Instead of using extra plane, we encode the number of tiles in x and y in the stride of + each planes (i.e. y_tiles << 16 | x_tiles) and introduce tile_mode, tile_width and + tile_height into GstVideoFormatInfo structure. + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2014-01-03 22:36:13 +0100 Wim Taymans <wtaymans@redhat.com> + + * docs/design/part-mediatype-video-raw.txt: + * gst-libs/gst/video/video-format.c: + * gst-libs/gst/video/video-format.h: + * gst-libs/gst/video/video-info.c: + * tests/check/elements/videoscale.c: + video: rename NV12T -> NV12_64Z32 + Is a bit more descriptive and allows us to add more tiled types + later. + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2014-01-03 22:29:09 +0100 Nicolas Dufresne <nicolas.dufresne at collabora.co.uk> + + * gst-libs/gst/video/video-frame.c: + video-frame: scale vertical tiles based on subsampling + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2014-01-03 22:18:08 +0100 Nicolas Dufresne <nicolas.dufresne at collabora.co.uk> + + * gst-libs/gst/video/video-frame.c: + video-frame: fix tiled pixel stride + Pixel stride is per component, not per plane. We get the tile mode from + the pixelstride of the TILE component. + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2013-12-26 17:40:05 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/video-format.h: + format: improve docs + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2013-12-25 16:22:32 +0100 Wim Taymans <wtaymans@redhat.com> + + * tests/check/elements/videoscale.c: + tests: fix videoscale test for NV12T + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2013-12-25 16:06:43 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/video-format.c: + * gst-libs/gst/video/video-frame.c: + video-format: fix off-by-one for tiled coordinates + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2013-12-25 15:22:24 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/video-tile.h: + video-tile: improve docs + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2013-12-25 14:57:30 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/video-format.c: + video-format: use shifts when possible + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2013-12-25 14:23:04 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/video-format.h: + * gst-libs/gst/video/video-frame.c: + video-frame: fix copy of tiled formats + Add code to copy tiled planes. + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2013-12-25 14:11:57 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/Makefile.am: + * gst-libs/gst/video/video-format.c: + * gst-libs/gst/video/video-tile.c: + * gst-libs/gst/video/video-tile.h: + video-tile: add tile mode and helper functions + Move the tile helper functions to their own file. Make it possible to + make other tiling modes later. + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2013-12-20 21:27:46 +0100 Wim Taymans <wtaymans@redhat.com> + + * docs/design/part-mediatype-video-raw.txt: + * gst-libs/gst/video/video-format.c: + * gst-libs/gst/video/video-format.h: + * gst-libs/gst/video/video-info.c: + video: add NV12T support + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2013-12-19 16:11:50 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/video-format.h: + Add tiled color format support + https://bugzilla.gnome.org/show_bug.cgi?id=707361 + +2014-01-13 15:32:23 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/pbutils/encoding-profile.c: + encoding-profile: Fix typo in the docs + +2014-01-11 01:14:19 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/libs/videodecoder.c: + tests: videodecoder: check that segment events are not dropped + Adds a test that simulates a scenario where the first buffers after + a segment can't be decoded and the decoder asks for those frames + to be released. The videodecoder base class should make sure that + the events attached to those first buffers are pushed even if the + buffers aren't going to be. + https://bugzilla.gnome.org/show_bug.cgi?id=721835 + +2014-01-11 01:24:44 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: do not lose events when dropping frames + Events must be persisted after a frame is dropped to avoid + losing obligatory information for the stream. + https://bugzilla.gnome.org/show_bug.cgi?id=721835 + +2014-01-08 11:29:29 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/libs/videodecoder.c: + tests: videodecoder: add test for reverse playback + Checks that buffers are pushed backwards in reverse playback + https://bugzilla.gnome.org/show_bug.cgi?id=721666 + +2014-01-06 20:53:15 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: use new segment earlier for reverse playback + For reverse playback, the segment event will only be pushed when + the first buffer is actually pushed. But for decoding frames and storing + those into the list to be pushed the output_segment.rate value is used + to determine if it is forward or reverse playback. + In case a previous segment event (or none) is in use it will mistakenly + think it is doing forward playback and push the buffers immediatelly and + try to clip buffers based on an old segment (or an uninitialized one, leading + to an assertion) + This patch fixes this by copying the segment earlier if on reverse playback + https://bugzilla.gnome.org/show_bug.cgi?id=721666 + +2014-01-10 14:24:12 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst/videotestsrc/gstvideotestsrc.c: + videotestsrc: fix unit test breaking on duration query + The new switch caused breaks to not break of the main switch + anymore, causing fall through. + +2014-01-10 15:06:23 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/videoconvert/gstvideoconvertorc-dist.c: + * gst/videoconvert/gstvideoconvertorc-dist.h: + videoconvert: Update disted orc files once again + +2014-01-10 11:17:38 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-play.c: + tools: gst-play: add dot file dumping for pipeline graph debugging + +2014-01-10 11:17:04 +0000 Tim-Philipp Müller <tim@centricular.com> + + * ext/pango/gstbasetextoverlay.c: + textoverlay: don't leak GAP events + +2014-01-10 09:53:21 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> + + * gst/videotestsrc/gstvideotestsrc.c: + videotestsrc: do not set TIME duration when asked for another format + This fixes asserts in pipelines such as: + gst-launch-1.0 videotestsrc num-buffers=1000 ! x264enc ! h264parse ! \ + matroskamux name=mux ! filesink location=test.mkv + +2014-01-10 09:21:08 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/videoconvert/gstvideoconvertorc-dist.c: + * gst/videoconvert/gstvideoconvertorc-dist.h: + videoconvert: Update disted orc files + +2014-01-09 18:12:00 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/videoconvert/gstvideoconvertorc.orc: + * gst/videoconvert/videoconvert.c: + videoconvert: rework YUV->RGB fastpaths + Rework the orc code to be around 10% faster and support arbitrary matrices. + Pass the matrix parameters to the YUV->RGB functions to make them work + for all matrices. This enables more and faster fastpath conversions. + See https://bugzilla.gnome.org/show_bug.cgi?id=721701 + +2014-01-09 18:08:41 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/videoconvert/gstvideoconvertorc.orc: + videoconvert: fix I420 to BGRA fast-path some more + Calculate alpha value differently so that we can avoid running out + of registers. + +2014-01-08 16:20:12 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/videoconvert/gstvideoconvertorc.orc: + videoconvert: remove unused code + +2014-01-03 15:24:29 +0100 Nicola Murino <nicola.murino@gmail.com> + + * gst-libs/gst/riff/riff-ids.h: + * gst-libs/gst/riff/riff-media.c: + riff: Add G726 ADPCM support + https://bugzilla.gnome.org/show_bug.cgi?id=720995 + +2014-01-07 22:04:20 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/libs/videodecoder.c: + tests: videodecoder: add check for serialization of events + Tests that events are properly serialized with buffers, also checks + that the usual events are sent (stream start, caps, segment and eos). + +2014-01-07 16:28:18 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * tests/check/Makefile.am: + * tests/check/libs/.gitignore: + * tests/check/libs/videodecoder.c: + tests: videodecoder: add simple playback test + Add a simple playback test that makes sure that video decoder pushes + buffers in the same order it receives and that it respects the + set timestamps and durations + +2014-01-07 15:01:14 +0100 Wim Taymans <wtaymans@redhat.com> + + * win32/common/libgstrtsp.def: + defs: update for new symbols + +2014-01-07 14:46:05 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/rtsp/gstrtsptransport.c: + rtsptransport: calculate default lower transport + Add an internal method to calculate the default lower transport whan it + is missing. + +2014-01-07 14:31:09 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/rtsp/gstrtsptransport.c: + * gst-libs/gst/rtsp/gstrtsptransport.h: + rtsptransport: add method to get media-type from transport + Add a method to make a media-type from the transport. Deprecate the old + method that only used the mode. + Based on patch from Aleix Conchillo Flaqué <aleix@oblong.com> + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720219 + +2014-01-07 11:51:01 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/rtsp/gstrtsptransport.c: + * gst-libs/gst/rtsp/gstrtsptransport.h: + rtsptransport: add GType for Profile + See https://bugzilla.gnome.org/show_bug.cgi?id=720696 + +2014-01-05 23:35:52 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/pbutils/descriptions.c: + * gst/typefind/gsttypefindfunctions.c: + typefind: add support of BWF RF64 a 64bit wav variant + Detect and describe the RF64 Broadcast Wave Format. + Fixes #519220 + +2014-01-05 21:39:52 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/riff/riff-read.c: + * gst-libs/gst/riff/riff-read.h: + * win32/common/libgstriff.def: + riff: remove new parse_ncdt api again + This chunk is avi specific, no need to expose this as public api. + +2014-01-04 22:30:17 +0100 Stefan Sauer <ensonic@users.sf.net> + + * win32/common/libgstriff.def: + win32: export new riff api + +2014-01-04 21:54:10 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/riff/riff-read.c: + riff: fix indentation messup from previous commit + +2014-01-04 21:31:07 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/riff/riff-ids.h: + * gst-libs/gst/riff/riff-read.c: + * gst-libs/gst/riff/riff-read.h: + riff: add support for nikon tags + Nikon cameras store metadata in a custom format. Add parsing of the chunk and + extract some initial data. + API: gst_riff_parse_ncdt() + Fixes #636143 + +2014-01-03 02:18:20 +1100 Jan Schmidt <jan@centricular.com> + + * gst-libs/gst/audio/gstaudiobasesrc.c: + audiobasesrc: Avoid unnecessary configuration + Port a change from audiobasesink from def07410, to ignore setcaps + when the caps don't actually change, and avoid a reconfiguration + and reset of the ringbuffer in that case. + +2013-11-15 14:17:03 +0000 William Grant <wgrant@ubuntu.com> + + * configure.ac: + configure: Prevent the NEON check in configure from passing under aarch64. + The test verifies that the NEON C intrinsics work, but the rest of the + codebase uses lots of direct ARMv7 NEON assembly. The same intrinsics + work in A64, but the assembly is slightly different. + Prevent the check from passing so that we don't use this where it won't + work. + https://bugzilla.gnome.org/show_bug.cgi?id=712367 + +2013-12-31 10:17:55 +0100 Stéphane Cerveau <scerveau@gmail.com> + + * gst-libs/gst/riff/riff-ids.h: + riff: Add id3 tag + Add id3 tag for wavparse + https://bugzilla.gnome.org/show_bug.cgi?id=721241 + +2013-12-31 09:37:36 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/icles/test-effect-switch.c: + Revert "test-effect-switch: Change one of the pad blocks to and idle probe" + This reverts commit 40fe5dcc84ff2cc7dbe0112d7830a33fd764d4e1. + Using an idle probe here is not ideal because we'll send an EOS event + from the application thread... which might block for quite some time. + Go back to a block probe. + +2013-12-30 19:48:29 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/videotestsrc/gstvideotestsrc.c: + videotestsrc: Always set pixel-aspect-ratio and interlace-mode in the fixed caps + Otherwise our caps will not be compatible with elements that require a + 1/1 pixel-aspect-ratio or progressive video. + https://bugzilla.gnome.org/show_bug.cgi?id=721103 + +2013-12-30 19:40:29 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/icles/test-effect-switch.c: + test-effect-switch: Don't put two format fields into the first capsfilter + +2013-12-30 19:12:53 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/icles/test-effect-switch.c: + test-effect-switch: Change one of the pad blocks to and idle probe + Just because we can. + +2013-12-30 17:30:15 +0100 Edward Hervey <bilboed@bilboed.com> + + * gst-libs/gst/pbutils/encoding-profile.c: + encoding-profile: Add missing break statement + And do a minor cleanup + COVERITY CID 1139753 + +2013-12-30 14:30:23 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/riff/riff-ids.h: + riff: add two chunk-ids for samples instruments + Wav files can have 'smpl' and 'inst' chunks. + +2013-12-30 13:46:34 +0100 Edward Hervey <bilboed@bilboed.com> + + * gst-libs/gst/riff/riff-media.c: + riff-media: Fix array read + nbchannels ranges from 1 to 8, therefore use '- 1' to get the proper + array value. + +2013-12-30 13:33:00 +0100 Edward Hervey <bilboed@bilboed.com> + + * gst/videorate/gstvideorate.c: + videorate: Remove useless assignement + Was already set before + +2013-12-26 17:47:46 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + gstrtpbasepayload: use the session's suggested ssrc after a collision, if the session provides one + Conflicts: + gst-libs/gst/rtp/gstrtpbasepayload.c + +2013-12-10 15:19:14 +0000 Matthieu Bouron <matthieu.bouron@collabora.com> + + * gst/playback/gstplaybin2.c: + * gst/playback/gstrawcaps.h: + playback: add ANY caps features to default audio/video raw caps + Allows elements using audio/video caps features to be used by playbin. + +2013-12-30 10:53:24 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/audio-info.c: + * gst-libs/gst/video/video-info.c: + audio/video-info: Properly initialize the info structures in set_format() + And don't assume in other code that set_format() preserves any fields at + all. These assumptions were already made here for fields that were changed + by set_format(). + +2013-12-30 10:14:09 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/audio-info.c: + * gst-libs/gst/video/video-info.c: + audio/video-info: Initialize the complete struct to 0 in the beginning + Instead of only initializing some parts in some code paths. Also + makes it easier to use the reserved bits of the structs later. + https://bugzilla.gnome.org/show_bug.cgi?id=720810 + +2013-12-20 19:48:06 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesrc.c: + audiobasesrc: Bunch of cosmetic/grammar fixes + +2013-12-20 18:58:43 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesrc.c: + audiobasesrc: Retarget FIXME to 2.0 + Properly fixing this one would break API. + +2013-12-20 18:54:39 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/audio.c: + * gst-libs/gst/audio/gstaudiobasesrc.c: + * gst-libs/gst/audio/gstaudiocdsrc.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/audio/gstaudioringbuffer.c: + * gst-libs/gst/audio/gstaudiosink.c: + * gst-libs/gst/audio/gstaudiosrc.c: + audiobase*: Drop trailing withespaces + +2013-12-20 18:53:13 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesrc.c: + audiobasesrc: Break some too long lines + +2013-12-20 18:41:59 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesrc.c: + audiobasesrc: Add FIXME for times in NSECONDS + Timebase is in nanoseconds pretty much everywhere else + +2013-12-26 23:21:45 +1100 Jan Schmidt <jan@centricular.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: Choose a default initial caps before sending GAP + If there are no caps from the audio decoder when handling a GAP + event - as when one is received right at the start on a DVD without + initial audio - then choose any default caps for downstream and + then send the GAP, so the audio sink has a configured format in + which to start the ringbuffer. + Also, make the audio sink reject a GAP without caps with a clearer + error message. + Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=603921 + +2013-12-26 17:41:00 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/rtsp/gstrtsptransport.c: + * gst-libs/gst/rtsp/gstrtsptransport.h: + rtsptransport: add more profiles + Add support for Feedback profiles + +2013-12-25 10:45:11 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/video/video-frame.c: + video-frame: fix plane copy for index plane + Move the code to handle the index plane in the _copy_plane. + +2013-12-24 01:20:25 +0000 Lionel Landwerlin <llandwerlin@gmail.com> + + * gst-libs/gst/video/colorbalance.c: + colorbalance: add missing annotation for list_channels() + https://bugzilla.gnome.org/show_bug.cgi?id=720999 + +2013-12-23 14:54:02 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/videoconvert/gstvideoconvertorc.orc: + * gst/videoconvert/videoconvert.c: + videoconvert: Fix I420 to BGRA fast-path alpha setting + This fast-path was adding 128 to every component including + alpha while it should only be done for all components except + alpha. This caused wrong alpha values to be generated. + Also remove the high-quality I420 to BGRA fast-path as it needs + the same fix, which causes an additional instruction, which causes + orc to emit more than 96 variables, which then just crashes. + This can only be fixed in orc by breaking ABI and allowing more + variables. + +2013-12-22 22:33:26 +0000 Tim-Philipp Müller <tim@centricular.com> + + * autogen.sh: + * common: + Automatic update of common submodule + From dbedaa0 to d48bed3 + +2013-12-22 21:56:03 +0000 Tim-Philipp Müller <tim@centricular.com> + + * po/Makevars: + po: set gettext domain in Makevars so we don't have to patch the generated Makefile.in.in + https://bugzilla.gnome.org/show_bug.cgi?id=705455 + +2013-12-22 22:07:43 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/libs/.gitignore: + tests: make git ignore new test binary + +2013-12-20 18:06:25 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + gstaudiobasesink: Always reset last_align + Should be done for all the reset_sync() cases. Not + only for the READY to PAUSED one. + +2013-12-20 18:02:42 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + gstaudiobasesink: Reset last_align to 0, not -1 + This is the expected behavior in READY -> PAUSED + +2013-12-20 17:58:43 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + gstaudiobasesink: Always reset avg_skew on _reset + Only case in which it wasn't (READY to PAUSED) should + have had this value reseted too. + +2013-12-20 17:10:44 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + gstaudiobasesink: Retarget FIXME to 2.0 + Properly fixing this one would break API + +2013-12-20 15:13:54 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + gstaudiobasesink: Factor out reset sync routine + +2013-12-20 01:06:33 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + gstaudiobasesink: Drop dead _sink_async_play() code + +2013-12-20 01:03:14 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + gstaudiobasesink: Break some too long lines + +2013-12-20 00:09:22 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + gstaudiobasesink: Cosmetics, grammar/spelling + - Drop repeated 'yet' from debug msg + - Drop repeated 'to' from param desc + - Some spelling + +2013-12-20 08:41:45 -0500 Edward Hervey <edward@collabora.com> + + * gst-libs/gst/audio/audio-info.c: + * gst-libs/gst/video/video-info.c: + audio/video: Initialize all {audio|video}info fields + Fixes "Unitialized Scalar Variable" issues reported by Coverity. + Has the added advantage of detecting whether somebody *does* use those + fields (ending up with a invalid address). + https://bugzilla.gnome.org/show_bug.cgi?id=720810 + +2013-12-19 17:41:31 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + gstaudiobasesink: Refactor alignment computation for clarity + +2013-12-18 15:52:09 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/subparse.c: + subparse: Add unit test for LRC subtitles + +2013-12-18 15:24:02 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/subparse/gstsubparse.c: + subparse: Add support for parsing LRC subtitles + https://bugzilla.gnome.org/show_bug.cgi?id=678590 + +2013-12-18 15:07:47 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/subparse/gstsubparse.c: + * gst/subparse/gstsubparse.h: + subparse: Add typefinder for LRC subtitles + +2013-12-10 13:54:28 -0800 Aleix Conchillo Flaqué <aleix@oblong.com> + + sdp: parse encryption key field + * gst-libs/gst/sdp/gstsdpmessage.c: parse encryption key field (k). + https://bugzilla.gnome.org/show_bug.cgi?id=720215 + +2013-12-17 18:04:33 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/pbutils/descriptions.c: + * gst/typefind/gsttypefindfunctions.c: + * tests/check/libs/pbutils.c: + pbutils: add typefinder and descriptions for audio/x-xi + xi files can be read by libsndfile. + +2013-12-17 18:03:40 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/pbutils/descriptions.c: + descriptions: longer version of two audio codec descriptions + +2013-12-17 17:25:07 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/video-format.h: + video-format: Document usage of GST_VIDEO_FORMAT_ENCODED + This must only ever be used in caps in combination with a non-system + memory GstCapsFeatures, and where it does not make sense to specify + any of the other video formats. Examples of this would be in gst-vaapi. + +2013-12-17 17:23:19 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/video-format.h: + * gst-libs/gst/video/video-info.c: + Revert "video: specify/restrict usage of GST_VIDEO_FORMAT_ENCODED" + This reverts commit 5fcdabd907ca45595b64131bbae0ea963e259a7c. + Instead of making it impossible to use the ENCODED format we should + just document that it must not be used for capsfeature-less caps. + Also this commit broke API/ABI. + +2013-12-17 17:09:02 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideoencoder.c: + videoencoder: Release the allocator on hard resets + +2013-12-16 15:53:41 +0000 Julien Isorce <julien.isorce@collabora.co.uk> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: release buffer pool and allocator on full reset + It allows to release the buffer pool sooner (i.e. when going + to GST_STATE_READY). Previously it was released in finalize. + Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=720389 + +2013-12-15 21:01:42 -0800 Todd Agulnick <todd@agulnick.com> + + * gst-libs/gst/audio/audio-format.c: + * sys/xvimage/xvimagesink.c: + Some compiler warning fixes to satisfy XCode compiler + https://bugzilla.gnome.org/show_bug.cgi?id=720513 + +2013-12-16 11:35:12 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/tag/gstvorbistag.c: + vorbistag: Read image-type from the GstSample info struct + But for backwards compatibility keep reading it from the caps and only + use the info struct if the caps don't contain the image-type. + +2013-12-13 14:36:41 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: gst_video_decoder_release_frame() is available since 1.2.2 + +2013-12-13 10:06:25 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-play.c: + tools: play: allow parse-launch strings for audio and video sink + +2013-12-12 13:42:59 +0100 Julien Isorce <julien.isorce@collabora.co.uk> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: change SSRC on GstRTPCollision event + Change our SSRC and update the caps when we receive a GstRTPCollision + event from downstream. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711560 + +2013-12-12 13:06:30 +0100 Julien Isorce <julien.isorce@collabora.co.uk> + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: implement src_event function + Add a srcpad event handler and call the src_event vmethod. + +2013-12-11 16:49:35 +0100 Edward Hervey <bilboed@bilboed.com> + + * gst-libs/gst/video/video-format.h: + * gst-libs/gst/video/video-info.c: + video: specify/restrict usage of GST_VIDEO_FORMAT_ENCODED + GST_VIDEO_FORMAT_ENCODED was added to support *extracting* video-related + information (like width, height, framerate,...) from caps. + It is __NOT__ intended to be used as a format field on video/x-raw caps. + +2013-12-10 00:13:55 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * tests/check/Makefile.am: + * tests/check/libs/rtp-basepayloading.c: + tests: Add test for rtpbasepayload/-depayload + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720162 + +2013-12-10 00:56:07 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * gst-libs/gst/rtp/gstrtpbuffer.c: + * tests/check/libs/rtp.c: + rtpbuffer: Allow subbuffering of empty buffers + See https://bugzilla.gnome.org/show_bug.cgi?id=720162 + +2013-12-09 16:34:22 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/convertframe.c: + convertframe: Fix indention + +2013-12-09 16:33:40 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideoencoder.c: + * gst-libs/gst/video/gstvideoencoder.h: + videoencoder: Add sink_query() src_query() virtual functions + Based on the videodecoder change by Nicolas Dufresne and applied + here for consistency. + https://bugzilla.gnome.org/show_bug.cgi?id=720103 + +2013-11-27 16:39:52 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * gst-libs/gst/video/gstvideodecoder.c: + * gst-libs/gst/video/gstvideodecoder.h: + videodecoder: Add sink_query() src_query() virtual + https://bugzilla.gnome.org/show_bug.cgi?id=720103 + +2013-12-09 13:55:28 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-play-kb.c: + tools: play: fix compiler warning on windows + +2013-12-06 19:27:04 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst-libs/gst/video/gstvideoutils.h: + videocodecframe: Correct function name in doc + +2013-12-06 16:23:46 -0500 Olivier Crête <olivier.crete@collabora.com> + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/video/gstvideoencoder.h: + videoencoder: Remove gst_video_encoder_set/get_discont + They've never existed outside the header file. + +2013-12-04 01:08:13 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * docs/design/Makefile.am: + docs: add missing files for distribution + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720015 + +2013-12-05 16:17:22 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst-libs/gst/audio/gstaudiobasesink.c: + audiobasesink: handle the RESYNC flag + Also resync when a buffer with the RESYNC flag is seen. + +2013-12-05 14:39:57 +0000 Julien Isorce <julien.isorce@collabora.co.uk> + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudioencoder.c: + audiodec/enc: clear reconfigure flag if negotiate succeeds + So that it avoids to send an allocation query twice. + One from an early call to gst_audio_encoder_negotiate from a + subclass, then one from gst_audio_encoder_allocate_output_buffer. + Which means that previously gst_audio_encoder_negotiate was not + clearing the GST_PAD_FLAG_NEED_RECONFIGURE even on success. + Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=719684 + +2013-12-05 14:31:25 +0000 Julien Isorce <julien.isorce@collabora.co.uk> + + * gst-libs/gst/video/gstvideodecoder.c: + * gst-libs/gst/video/gstvideoencoder.c: + videodec/enc: clear reconfigure flag if negotiate succeeds + So that it avoids to send an allocation query twice. + One from an early call to gst_video_encoder_negotiate from a + subclass, then one from gst_video_encoder_allocate_output_frame. + Which means that previously gst_video_encoder_negotiate was not + clearing the GST_PAD_FLAG_NEED_RECONFIGURE even on success. + Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=719684 + +2013-12-05 11:39:07 +0100 Sebastian Dröge <sebastian@centricular.com> + + * ext/theora/gsttheoradec.c: + theoradec: Use new gst_video_decoder_set_needs_format() API + +2013-12-05 11:37:09 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: Use FALSE instead of 0 + +2013-12-05 11:34:36 +0100 Sebastian Dröge <sebastian@centricular.com> + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/video/gstvideodecoder.c: + * gst-libs/gst/video/gstvideodecoder.h: + * win32/common/libgstvideo.def: + videodecoder: Add API to allow subclasses to specify that they needs caps before any buffers + +2013-12-05 11:25:47 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/video/gstvideoencoder.c: + videoencoder: Return not-negotiated if we don't have caps when the first buffer arrives + Otherwise things like filesrc ! jpegenc ! fakesink just crash with + a segmentation fault because subclasses expect caps to be there. + +2013-12-04 19:24:08 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: no fallback to segment start for reverse playback + See https://bugzilla.gnome.org/show_bug.cgi?id=709965 + +2013-12-05 00:27:14 +0900 Justin Joy <justin.joy.9to5@gmail.com> + + * gst-libs/gst/video/convertframe.c: + convertframe: Fix trivial memory leak in debug statement + gst_element_get_name() requires the caller to g_free() the return value + https://bugzilla.gnome.org/show_bug.cgi?id=719850 + +2013-12-02 20:35:04 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: use segment start as fallback ts if no other available + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709965 + +2013-12-01 12:37:52 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net> + + * docs/libs/gst-plugins-base-libs-sections.txt: + * win32/common/libgstvideo.def: + videodecoder: add new API to docs and defs + +2013-11-26 20:50:33 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net> + + * gst-libs/gst/video/gstvideodecoder.c: + * gst-libs/gst/video/gstvideodecoder.h: + videodecoder: make _release_frame external API + ... so subclasses can release a frame all the way (also from frame list) + without having to pass through _finish_frame or _drop_frame. + The latter may not be applicable, or may or may not have already + been called for the frame in question. + See https://bugzilla.gnome.org/show_bug.cgi?id=693772 + +2013-11-26 20:51:58 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: fix spelling error in debug message + +2013-11-29 17:30:09 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/playback/gsturidecodebin.c: + uridecodebin: copy sticky events + +2013-11-29 17:26:13 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/playback/gstdecodebin2.c: + decodebin2: copy sticky events + +2013-11-29 13:32:55 +0100 Sebastian Dröge <sebastian@centricular.com> + + * ext/theora/gsttheoraparse.c: + theoraparse: Fix event handling + Send CAPS event before any SEGMENT events or any other events + that must come in order after the CAPS event. + +2013-11-29 09:04:20 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-play.c: + tools: gst-play: quit on Q or Esc key + +2013-11-28 16:22:01 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/tcp/gsttcpserversink.c: + tcp: fix compilation with MSVC + error C2440 at line 165 of gsttcpserversink.c + type cast error: cannot convert from GSocket* to GstMultiSinkHandle + +2013-11-28 11:25:20 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/playback/gstdecodebin2.c: + decodebin2: activate ghost pad before targetting + Activate the decodebin2 pad before setting the target. This makes sure + that the events are copied. + +2013-11-21 22:54:42 +1100 Matthew Waters <ystreet00@gmail.com> + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/video/gstvideometa.h: + videometa: add GstVideoGLTextureUploadMeta buffer pool option + allows configuration of whether GstVideoGLTextureUploadMeta is + added to buffers resulting from a buffer pool. This is sperate + to the caps feature in that an element may want to add the upload + meta itself rather than allowing the buffer pool to. + https://bugzilla.gnome.org/show_bug.cgi?id=712798 + +2013-11-26 12:29:30 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: error out if no frames are decoded before eos + Raise an error in case no frames are decoded before EOS and we + have input, meaning that data was received but it was somehow invalid. + Based on the videodecoder change, merged here for consistency. + https://bugzilla.gnome.org/show_bug.cgi?id=711094 + +2013-11-26 12:20:33 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: Allow using -1 for infinite tolerated errors + Allows using -1 to make audiodecoder never post an error message + after decoding errors. + Based on the videodecoder change, merged here for consistency. + https://bugzilla.gnome.org/show_bug.cgi?id=711094 + +2013-11-26 12:03:24 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/playback/gstplaysink.c: + playsink: Fix visualizations if no visualization plugin was set + https://bugzilla.gnome.org/show_bug.cgi?id=712280 + +2013-10-29 14:40:23 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: error out if no frames are decoded before eos + Raise an error in case no frames are decoded before EOS and we + have input, meaning that data was received but it was somehow invalid. + https://bugzilla.gnome.org/show_bug.cgi?id=711094 + +2013-10-29 14:11:51 -0300 Thiago Santos <ts.santos@sisa.samsung.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: allow using -1 for infinite tolerated errors + Allows using -1 to make videodecoder never post an error message + after decoding errors. + https://bugzilla.gnome.org/show_bug.cgi?id=711094 + +2013-11-24 14:38:25 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-play-kb.h: + * tools/gst-play.c: + tools: play: implement seeking via console in interactive mode + Arrow left and right to seek back of forward. + +2013-11-24 14:33:24 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-play.c: + tools: play: fix endless loop on unhandled keys + When debugging output is not enabled. + +2013-11-24 13:49:04 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-play.c: + tools: play: add keyboard controls for next/previous item in list + Make the '>' and '<' keys skip to the next or previous item in + the playlist. + +2013-11-24 01:08:48 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tools/Makefile.am: + * tools/gst-play-kb.c: + * tools/gst-play-kb.h: + * tools/gst-play.c: + tools: play: add --interactive switch and basic keyboard handling + Only pause/play with spacebar for now. + +2013-11-23 11:25:28 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/typefind/gsttypefindfunctions.c: + typefind: Add typefinder for OpenEXR + +2013-11-21 21:33:59 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: avoid descending output timestamps + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712796 + +2013-11-22 21:00:21 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tools/gst-play.c: + tools: play: add --shuffle command line option + +2013-11-21 16:34:25 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/elements/subparse.c: + tests: add unit test for samiparser issue + https://bugzilla.gnome.org/show_bug.cgi?id=712805 + +2013-11-21 22:04:46 +0900 Jihyun Cho <jihyun.jo@gmail.com> + + * gst/subparse/samiparse.c: + subparse: fix null pointer access in sami parser + https://bugzilla.gnome.org/show_bug.cgi?id=712805 + +2013-11-21 15:19:47 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/subparse/gstssaparse.c: + * gst/subparse/gstsubparse.c: + subparse: g_memmove() is deprecated + Just use plain memmove(), g_memmove() is deprecated in + recent GLib versions. + https://bugzilla.gnome.org/show_bug.cgi?id=712811 + +2013-11-18 19:27:14 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tests/icles/input-selector-test.c: + tests: fix input-selector-test + Update for pad template name changes. + +2013-11-18 16:03:07 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/elements/appsrc.c: + tests: fix appsrc test with latest GLib version + With the latest GLib, g_source_remove() complains about not finding + the timeout source with the given ID here, since it was already + destroyed by returning FALSE from the timeout callback. Also return + FALSE from the bus watches when we don't want to be called any more. + +2013-11-16 13:06:37 +0000 Tim-Philipp Müller <tim@centricular.com> + + * ext/cdparanoia/gstcdparanoiasrc.c: + * ext/pango/gstbasetextoverlay.c: + * ext/theora/gsttheoraparse.c: + * gst/app/gstapp.c: + * gst/audiorate/gstaudiorate.c: + * gst/gio/gstgiosink.c: + * gst/gio/gstgiosrc.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultihandlesink.c: + * gst/tcp/gstmultioutputsink.c: + * gst/tcp/gstmultisocketsink.c: + * gst/videorate/gstvideorate.c: + * sys/ximage/ximagesink.c: + * sys/xvimage/xvimagesink.c: + docs: remove old 0.10 Since markers + They're just confusing. + +2013-11-16 12:29:04 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtspdefs.c: + * gst-libs/gst/rtsp/gstrtsprange.c: + * gst-libs/gst/rtsp/gstrtsprange.h: + docs: cosmetic since marker fixes + +2013-11-16 15:24:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net> + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: also set output buffer DTS + +2013-11-14 01:53:31 -0300 Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com> + + * gst/typefind/gsttypefindfunctions.c: + typefind: Fix identification of some MPEG files + Make sure we begin by peeking at MPEG2_MAX_PROBE_LENGTH + bytes. + Fixes: + https://bugzilla.gnome.org/show_bug.cgi?id=678011 + +2013-11-13 20:12:48 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/rtp/gstrtpbuffer.c: + rtpbuffer: Fix gst_rtp_buffer_ext_timestamp() with clang 5 on iOS/ARM + The bitwise NOT operator is not defined on signed integers. + Thanks to Wim Taymans for finding the cause. + https://bugzilla.gnome.org/show_bug.cgi?id=711819 + +2013-11-12 18:58:43 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/elements/streamsynchronizer.c: + tests: fix race in streamsynchronizer test + Wait for thread to exit before starting to free the + to_push list, otherwise thread might check the final + to_push->next node only after we've freed it already. + +2013-11-11 14:10:53 +0200 Sreerenj Balachandran <sreerenj.balachandran@intel.com> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: try to negotiate the buffer pool even though there is no o/p format + We could have allocation query before caps event and even without caps inside + the query. In such cases , the downstream can return a bufferpool object with + out actually configuring it. This feature is helpful to negotiate the bufferpool + with out knowing the output video format. For eg: some hardware accelerated + decoders can interpret the o/p video format only after it finishes the decoding + of one buffer at least. + https://bugzilla.gnome.org/show_bug.cgi?id=687183 + +2013-11-07 15:03:34 +0000 Tom Greenwood <tcdgreenwood@hotmail.com> + + * gst-libs/gst/app/gstappsrc.c: + appsrc: Fix deadlock that may occur when multiple threads access appsrc at once + https://bugzilla.gnome.org/show_bug.cgi?id=711550 + +2013-11-04 09:55:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst-libs/gst/tag/gsttagdemux.c: + tagdemux: accumulate buffers in adapter + Accumulate buffers in an adapter instead of appending them because append causes + a lot of memcpys. + Keep track of the last tagsize and accumulate enough data before attempting to + parse more data. + This patch implements a minimal amount of changes in order to not change the + behaviour. We should really rewrite the tag handling and trimming using + the adapter API instead of merging and trimming into a buffer. + +2013-11-06 12:16:31 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/adder.c: + adder: Free consistency checker instance in test_live_seeking test + +2013-11-06 12:01:14 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/adder.c: + adder: Release some request pads properly in the unit test + +2013-11-05 11:18:01 +0000 Tim-Philipp Müller <tim@centricular.com> + + * common: + Automatic update of common submodule + From 865aa20 to dbedaa0 + +2013-11-04 11:34:38 +0100 Alessandro Decina <alessandro.d@gmail.com> + + * tools/gst-discoverer.c: + discoverer: fix build after last commit + Add a forward declaration for my_g_string_append_printf that specifies + G_GNUC_PRINTF. Turn off indent on it as it drives gst-indent crazy. + +2013-11-04 11:17:30 +0100 Alessandro Decina <alessandro.d@gmail.com> + + * tools/gst-discoverer.c: + discoverer: fix -Wformat-nonliteral warning + +2013-11-03 15:57:54 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/libs/audio.c: + audio: Add unit test for filling memory with silence samples + +2013-11-03 12:23:12 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/gstaudiopack-dist.c: + * gst-libs/gst/audio/gstaudiopack-dist.h: + audio: Update ORC dist files + +2013-11-03 12:22:33 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/audio/audio-format.c: + * gst-libs/gst/audio/gstaudiopack.orc: + audio-format: Use ORC for filling memory with silence samples + +2013-11-01 17:02:22 +0100 Sebastian Dröge <sebastian@centricular.com> + + * docs/libs/gst-plugins-base-libs-sections.txt: + * win32/common/libgstrtsp.def: + rtspconnection: Add new API to the docs and .def file + +2013-11-01 16:43:56 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst-libs/gst/rtsp/gstrtspconnection.h: + rtspconnection: Fix indention in header + +2013-11-01 07:25:01 -0700 Aleix Conchillo Flaque <aleix@oblong.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtspconnection.h: + rtspconnection: allow setting tls certificate validation + Added new functions gst_rtsp_connection_set_tls_validation_flags() to + allow setting the TLS certificate validation flags when establishing a + TLS connection. + A getter is also available, gst_rtsp_connection_get_tls_validation_flags(). + https://bugzilla.gnome.org/show_bug.cgi?id=711231 + +2013-11-01 14:22:13 +0000 Matthieu Bouron <matthieu.bouron@collabora.com> + + * gst-libs/gst/sdp/gstsdpmessage.c: + sdp: fix duplicate 'const' declaration warnings + https://bugzilla.gnome.org/show_bug.cgi?id=711258 + +2013-10-16 16:46:05 -0300 Thibault Saunier <thibault.saunier@collabora.com> + + * gst/playback/gstrawcaps.h: + playback: Add subpicture/x-dvb as raw caps + https://bugzilla.gnome.org/show_bug.cgi?id=710325 + +2013-10-28 12:36:04 +0100 Antonio Ospite <ospite@studenti.unina.it> + + * gst/videoscale/gstvideoscale.c: + videoscale: fix adding borders when NV12 is used + When the frame buffer is NV12 the borders are not added at all, fix that + and fill them to black. + https://bugzilla.gnome.org/show_bug.cgi?id=711003 + +2013-10-23 16:43:32 +0100 Matthieu Bouron <matthieu.bouron@gmail.com> + + * gst/videoconvert/videoconvert.c: + videoconvert: remove unneeded guint comparaison + https://bugzilla.gnome.org/show_bug.cgi?id=710760 + +2013-10-14 18:45:16 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: also filter 'framed' field when looking for same streams + Fixes extra streams for some mp4 files containing aac audio. + +2013-10-08 21:57:11 +0200 Stefan Sauer <ensonic@users.sf.net> + + * ext/ogg/gstoggdemux.c: + oggdemux: fix copy'n'paste in comment + +2013-10-10 15:56:32 -0300 Thibault Saunier <thibault.saunier@collabora.com> + + * ext/theora/gsttheoraenc.c: + theoraenc: Do nothing when flushing the encoder when no caps were set + In case we receive a flush event before having our caps set, we will + end up trying to create a theora encoder even though we are not ready. + Avoid that situation making sure we are initialized before accepting to + be flushed. + https://bugzilla.gnome.org/show_bug.cgi?id=709858 + +2013-10-11 21:51:00 +0200 Stephan Sundermann <stephansundermann@gmail.com> + + * gst-libs/gst/video/navigation.c: + navigation: Add missing out parameter annotations to GstNavigation + https://bugzilla.gnome.org/show_bug.cgi?id=709938 + +2013-10-10 14:09:19 +0100 Julien Isorce <julien.isorce@collabora.co.uk> + + * tests/examples/overlay/qtgv-videooverlay.cpp: + examples/overlay: handle the case when xvimagesink is not found + So that ximagesink can have a chance to be found. + In qtgv-videooverlay. + +2013-10-10 14:01:44 +0100 Julien Isorce <julien.isorce@collabora.co.uk> + + * tests/examples/overlay/gtk-videooverlay.c: + * tests/examples/overlay/qt-videooverlay.cpp: + examples/overlay: unref sink only when found + In gtk-videooverlay and qt-videooverlay examples. + +2013-10-07 14:52:00 -0300 Thibault Saunier <thibault.saunier@collabora.com> + + * gst-libs/gst/pbutils/encoding-profile.c: + * gst/encoding/gstencodebin.c: + encodebin: Handle changes in encoding_profile::restriction during playback + There are cases where we want to change the restrictions caps during + playback, handle that in encodebin. + https://bugzilla.gnome.org/show_bug.cgi?id=709588 + +2013-10-08 17:07:02 +0200 Takashi Iwai <tiwai@suse.de> + + * ext/alsa/gstalsa.c: + * ext/alsa/gstalsa.h: + * ext/alsa/gstalsasink.c: + * ext/alsa/gstalsasrc.c: + alsa: Add channel map API support + The initial support for the new ALSA chmap API. + Just translate the current chmap to GstAudioChannelPosition during the + setup. No function to specify the channel map manually yet, so still + impossible to assign any non-standard positions or to configure in a + different order even if the hardware allows. + https://bugzilla.gnome.org/show_bug.cgi?id=709755 + +2013-10-08 16:02:46 +0200 Takashi Iwai <tiwai@suse.de> + + * gst-libs/gst/audio/gstaudioringbuffer.c: + audioringbuffer: Don't clear need_reorder flag too early + gst_audio_ring_buffer_set_channel_positions() checks whether the given + positions are identical with the current setup and returns + immediately if so. But it also clears need_reorder flag before this + comparison, thus this flag might be wrongly cleared if the function is + called twice with the same channel positions. + Move the flag clearance after the check. + https://bugzilla.gnome.org/show_bug.cgi?id=709754 + +2013-10-08 16:13:58 -0300 Thiago Santos <ts.santos@partner.samsung.com> + + * tests/check/elements/videotestsrc.c: + videotestsrc: improve test for backwards playback + Improve test by checking that timestamps are decreasing + +2013-10-08 16:10:54 -0300 Thiago Santos <ts.santos@partner.samsung.com> + + * gst/videotestsrc/gstvideotestsrc.c: + * tests/check/elements/videotestsrc.c: + videotestsrc: implement duration query + Add duration query to videotestsrc, it can answer this query when + the num-buffers property is set. + https://bugzilla.gnome.org/show_bug.cgi?id=709646 + +2013-06-07 16:32:23 -0400 Thibault Saunier <thibault.saunier@collabora.com> + + * tests/check/elements/videotestsrc.c: + tests: test videotestsrc in reverse playback + https://bugzilla.gnome.org/show_bug.cgi?id=701813 + +2013-10-08 00:08:34 -0300 Thiago Santos <ts.santos@partner.samsung.com> + + * gst/videotestsrc/gstvideotestsrc.c: + * gst/videotestsrc/gstvideotestsrc.h: + videotestsrc: implement reverse playback + Decrement the n_frames counter when doing reverse playback to + have timestamps and offsets reducing instead of increasing + https://bugzilla.gnome.org/show_bug.cgi?id=701813 + +2013-10-08 09:13:50 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/video/gstvideodecoder.c: + videodecoder: don't overflow in bytes<->time conversion + fps_n and _d values can be large and this can overflow a uint. Also fix + copy'n'paste mistake in comments. + +2013-10-07 22:52:27 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: filter 'parsed' field when checking for same caps + We're checking the caps to see if we got more caps details after a parser got + plugged. This will also have a flipped 'parsed' field. If the field was already + present before the parse the match will fail. Add a function that will do the + check while excluding this field. + +2013-10-07 22:51:46 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: don't shadow local variables + +2013-10-07 22:51:04 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: early return when we have no streams + +2013-10-07 22:49:52 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: also log stream-id + +2013-10-07 18:53:18 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: fix quark-mismatch for toc and stream-id + Seems like a copy'n'paste from 15ee41df. + +2013-10-05 21:01:53 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: report depth for video + This was returning 0 in all cases. Use the data from GstVideoFormatInfo instead. + +2013-10-04 13:57:51 +0200 Matej Knopp <matej.knopp@gmail.com> + + * gst/audioconvert/gstaudioconvert.c: + audioconvert: Map buffer as READWRITE if the buffer and memory is writable + and only use the input buffer as temporary buffer in that case. + https://bugzilla.gnome.org/show_bug.cgi?id=709408 + +2013-09-30 21:46:10 +0200 Hans Månsson <hansm@axis.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: Connect to proxy if specified + Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708880 + +2013-10-03 19:52:58 +0200 Stefan Sauer <ensonic@users.sf.net> + + * tools/gst-discoverer.c: + discoverer: extract helper to print common stream info + Save some lnes of code by using a helper for common stream info. + +2013-10-02 11:27:41 +0200 Stefan Sauer <ensonic@users.sf.net> + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: extract some common code + Extract code to make a GstDiscovererInfo. Extracts code that sets StreamInfo. + +2013-10-02 15:02:44 +0200 Sebastian Dröge <slomo@circular-chaos.org> + + * gst/playback/gstplaysink.c: + playsink: If the visualisation is changing and reconfiguration is pending, do it all during reconfiguration + Otherwise we will have two pad blocks that want to use the same mutex + and block each other via the streamlock. + https://bugzilla.gnome.org/show_bug.cgi?id=709210 + +2013-10-02 13:06:03 +0200 Edward Hervey <edward@collabora.com> + + * win32/common/libgstpbutils.def: + win32: Update defs file + +2013-10-02 12:26:59 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com> + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/pbutils/codec-utils.c: + * gst-libs/gst/pbutils/codec-utils.h: + * win32/common/libgstpbutils.def: + pbutils: Add codec-utility funtions to support H265 + https://bugzilla.gnome.org/show_bug.cgi?id=708921 + +2013-10-01 23:17:06 +0200 Sebastian Dröge <slomo@circular-chaos.org> + + * gst-libs/gst/pbutils/descriptions.c: + descriptions: Add description for H.265 + +2013-09-24 15:51:46 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com> + + * gst/typefind/gsttypefindfunctions.c: + typefind: Add typefind function for H265 + https://bugzilla.gnome.org/show_bug.cgi?id=708680 + +2013-09-24 16:47:52 -0700 Thiago Santos <ts.santos@partner.samsung.com> + + * gst/playback/gstplaybin2.c: + playbin: make sure elements are in null before disposing + If a pipeline fails to preroll, it might happen that the sinks are + put into READY state from playbin's sink activation, but they are never + set to playsink, so they aren't being managed by a GstBin and will keep + their READY state until they are unreffed, leading to a warning. + Prevent this by always forcing them to NULL when deactivating a group + https://bugzilla.gnome.org/show_bug.cgi?id=708789 + +2013-09-28 13:19:02 +0200 Johannes Dewender <gnome@JonnyJD.net> + + * gst-libs/gst/audio/gstaudiocdsrc.c: + audiocdsrc: Don't consider trailing data tracks for MusicBrainz disc id calculation + MusicBrainz removes trailing data tracks from releases on the server + and also for the calculation of the MusicBrainz Disc ID. + https://bugzilla.gnome.org/show_bug.cgi?id=708991 + +2013-09-23 11:35:43 +0200 David Svensson Fors <davidsf@axis.com> + + * gst-libs/gst/audio/gstaudioringbuffer.c: + audioringbuffer: check if acquired in set_timestamp + Also use GST_OBJECT_LOCK when accessing object data in set_timestamp. + https://bugzilla.gnome.org/show_bug.cgi?id=702230 + +2013-09-15 21:48:43 +0200 MathieuDuponchelle <mathieu.duponchelle@epitech.eu> + + * gst/adder/gstadder.c: + adder: Don't take channel mask in consideration in mono or stereo + This could cause negotiation to fail. + https://bugzilla.gnome.org/show_bug.cgi?id=708633 + +2013-09-27 22:41:28 +0200 Matej Knopp <matej.knopp@gmail.com> + + * gst/audiorate/gstaudiorate.c: + audiorate: clip buffer before pushing it + https://bugzilla.gnome.org/show_bug.cgi?id=708953 + +2013-09-27 22:40:28 +0200 Matej Knopp <matej.knopp@gmail.com> + + * gst-libs/gst/audio/audio.c: + audio: change buffer timestamp when clipping even if data hasn't been trimmed + https://bugzilla.gnome.org/show_bug.cgi?id=708952 + +2013-09-27 22:53:43 +0200 Matej Knopp <matej.knopp@gmail.com> + + * gst-libs/gst/pbutils/descriptions.c: + pbutils: Add entry for text/x-raw + https://bugzilla.gnome.org/show_bug.cgi?id=708954 + +2013-09-25 19:29:24 +0200 Matej Knopp <matej.knopp@gmail.com> + + * gst-libs/gst/pbutils/descriptions.c: + pbutils: add MPEG 2 AAC description + https://bugzilla.gnome.org/show_bug.cgi?id=708773 + +2013-09-25 15:17:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst-libs/gst/audio/gstaudiobasesink.c: + audiobasesink: do big correction for large drift + If we are using skew slaving and we drift more than twice the allowed amount, do + a big correction to get back on track more quickly. + +2013-09-24 18:28:57 +0100 Tim-Philipp Müller <tim@centricular.net> + + * README: + * common: + Automatic update of common submodule + From 6b03ba7 to 865aa20 + +2013-09-24 16:26:37 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: Unset input/output_stream after freeing the GIOStream + watch->input_stream and watch->output_stream are owned by the GIOStream + and should be unset after freeing the stream. + https://bugzilla.gnome.org/show_bug.cgi?id=708689 + +2013-09-24 15:05:21 +0200 Sebastian Dröge <slomo@circular-chaos.org> + + * configure.ac: + configure: Actually use 1.3.0.1 as version to make configure happy + +2013-09-24 15:00:20 +0200 Sebastian Dröge <slomo@circular-chaos.org> + + * configure.ac: + Back to development + === release 1.2.0 === -2013-09-24 Sebastian Dröge <sebastian.droege@collabora.co.uk> +2013-09-24 14:16:22 +0200 Sebastian Dröge <slomo@circular-chaos.org> + * ChangeLog: + * NEWS: + * RELEASE: * configure.ac: - releasing 1.2.0 + * docs/plugins/inspect/plugin-adder.xml: + * docs/plugins/inspect/plugin-alsa.xml: + * docs/plugins/inspect/plugin-app.xml: + * docs/plugins/inspect/plugin-audioconvert.xml: + * docs/plugins/inspect/plugin-audiorate.xml: + * docs/plugins/inspect/plugin-audioresample.xml: + * docs/plugins/inspect/plugin-audiotestsrc.xml: + * docs/plugins/inspect/plugin-cdparanoia.xml: + * docs/plugins/inspect/plugin-encoding.xml: + * docs/plugins/inspect/plugin-gio.xml: + * docs/plugins/inspect/plugin-ivorbisdec.xml: + * docs/plugins/inspect/plugin-libvisual.xml: + * docs/plugins/inspect/plugin-ogg.xml: + * docs/plugins/inspect/plugin-pango.xml: + * docs/plugins/inspect/plugin-playback.xml: + * docs/plugins/inspect/plugin-subparse.xml: + * docs/plugins/inspect/plugin-tcp.xml: + * docs/plugins/inspect/plugin-theora.xml: + * docs/plugins/inspect/plugin-typefindfunctions.xml: + * docs/plugins/inspect/plugin-videoconvert.xml: + * docs/plugins/inspect/plugin-videorate.xml: + * docs/plugins/inspect/plugin-videoscale.xml: + * docs/plugins/inspect/plugin-videotestsrc.xml: + * docs/plugins/inspect/plugin-volume.xml: + * docs/plugins/inspect/plugin-vorbis.xml: + * docs/plugins/inspect/plugin-ximagesink.xml: + * docs/plugins/inspect/plugin-xvimagesink.xml: + * gst-plugins-base.doap: + * win32/common/_stdint.h: + * win32/common/config.h: + Release 1.2.0 + +2013-09-24 14:14:18 +0200 Sebastian Dröge <slomo@circular-chaos.org> + + * 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/hr.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: + Update .po files 2013-09-24 12:47:26 +0200 Sebastian Dröge <slomo@circular-chaos.org> @@ -1,123 +1,110 @@ -This is GStreamer Base Plugins 1.2.0 +This is GStreamer Base Plugins 1.3.1 -Changes since 1.0: +Changes since 1.2: New API: - • GstContext negotiation / sharing / announcing for sharing a - generic context between elements, e.g. a display handle - • GL texture upload conversion meta for allowing different - buffer types to be converted to an OpenGL texture - • GstCapsFeatures as extension to GstCaps for allowing the - negotiation of specific memory or meta requirements between - elements - • GstMemory flags for contiguous and non-mappable memory - • The stream-start event has optional flags now, e.g. for signalling - sparse streams - • The stream-start even has an optional group-id field now to signal - all streams that should be played together - • Allocators library in gst-plugins-base, currently only with generic - dmabuf memory support - • insertbin library for easier handling of dynamically linked - pipelines (in -bad for now) - • EGL helper library (in -bad for now) - • MPEG-TS data structure library (in -bad for now) - • New GstVideoRegionOfInterestMeta to describe a region of interest on - video frames. - • GstVideoDecoder/Encoder has new ::flush() vfunc to replace the - ill-defined ::reset() vfunc. - • The URI query allows to query the redirected URI now. + • GstMessageType has GST_MESSAGE_EXTENDED added. All types before + that can be used together as a flags type as before, but from + that message onwards the types are just counted incrementally. + This was necessary to be able to add more message types. + In 2.0 GstMessageType will just become an enum and not a flags + type anymore. + • GstDeviceMonitor for device probing, e.g. to list all available + audio or video capture devices. This is the replacement for + GstPropertyProbe from 0.10. + • Events accumulate the running-time offset now when travelling + through pads, as set by the gst_pad_set_offset() function. This + allows to compensate for this in the QOS event for example. + • GstBuffer has a new flag "tag-memory" that is set automatically + when memory is added or removed to a buffer. This allows buffer + pools to detect if they can recycle a buffer or need to reset + it first. + • GstToc has new API to mark GstTocEntries as loops. + • A not-authorized resource error has been defined to notify + applications that accessing the resource has failed because + of missing authorization and to distinguish this case from others. + This change is actually already in 1.2.4. + • GstPad has a new flag "accept-intersect", that will let the default + ACCEPT_CAPS query handler do an intersection instead of subset check. + This is interesting for parser elements that can handle incomplete + caps. + • GstCollectPads has support for flushing and a default handler for + SEEK events now. + • GstSegment has new API to offset the running time by a specific + value and this is used in GstPad to allow positive and negative + offsets in gst_pad_set_offset() in all situations. + • Support for h265/HEVC and VP8 has been added to the codec utils and codec + parsers library, and was integrated into various elements. + • API for adjusting the TLS validation of RTSP connection has been added. + • The RTSP and SDP library has MIKEY (RFC 3830) support now, and + there is API to distinguish between the different RTSP profiles. + • API to access RTP time information and statistics. + • Support for auxiliary streams was added to rtpbin. + • Support for tiled, raw video formats has been added. + • GstVideoDecoder and GstAudioDecoder have API to help aggregating tag + events and merge custom tags into them consistently. + • playbin/playsink has support for application provided audio and video + filters. + • The GL library was merged from gst-plugins-gl to gst-plugins-bad, + providing a generic infrastructure for handling GL inside GStreamer + pipelines and a plugin with some elements using these, especially + a video sink. Supported platforms currently are Android, Cocoa (OS X), + DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11, + Wayland and EGL platforms. + This replaces eglglessink and also is supposed to replace osxvideosink. -Major changes: - • New tool: gst-play-1.0 in gst-plugins-base for basic playback - testing on the command line. - • New plugins: - ∘ mssdemux for Microsoft Smooth Streaming - ∘ dashdemux for DASH adaptive streaming protocol - ∘ bluez for interaction with Bluetooth devices - ∘ openjpeg for JPEG2000 decoding and encoding - ∘ daala for experimental Daala decoding and encoding - ∘ vpx plugin has experimental VP9 decoding and encoding support - ∘ webp plugin for WebP decoding (encoding to be added later) - ∘ Various others: yadif, srtp, sbc, fluidsynth, midiparse, - mfc, ivtv, accuraterip and audiofxbad - - • Moved plugins: - ∘ dtmf, vp8rtp, scaletempo and rtpmux plugins are in - gst-plugins-good now - - • Video: - ∘ Fix handling of interlaced video in converters such as videoscale - and videoconvert (e.g. scale both fields independently) - ∘ videoconvert will try harder to minimise quality losses when - conversion is necessary - ∘ The experimental GstSurfaceConverter, GstSurfaceMeta and - GstVideoContext APIs from the (confusingly-named) - libgstbasevideo-1.0 library in gst-plugins-bad have now been - removed and been replaced by new APIs in GStreamer Core and - gst-plugins-base (see above). Since that was all that was left in - this library, the entire experimental libgstbasevideo-1.0 library - has been removed from gst-plugins-bad - ∘ Chroma subsampling and chroma siting conversion is better handled - in videoconvert and the support for interlaced video was improved. - ∘ New pinwheel and spoke patterns in videotestsrc - ∘ videomixer can now accept different video formats on its sinkpads - and converts to a common format during mixing - • Audio: - ∘ audioconvert will try harder to minimise quality losses when - conversion is necessary - ∘ adder now allows muting/unmuting of its input streams, and also - per-input stream volume - ∘ pulseaudio elements can switch between devices during playback now - ∘ aacparse can convert between ADTS←→RAW - - • Platform specific changes: - ∘ Caps, events, etc. are now printed in the GStreamer debug logs - with their content instead of just the pointer address even on - non-glibc platforms (e.g. Windows, OSX, Android). - ∘ Network elements (UDP/TCP) now work better with platforms, - where IPv6 sockets can't handle IPv4 (e.g. Windows) - ∘ Linux/BSD: v4l2 had many improvements and cleanups +Major changes: + • New plugins and elements: + ∘ v4l2videodec element for accessing hardware codecs on + platforms that make them accessible via V4L2, e.g. + Samsung Exynos. This comes together with major refactoring + of the existing V4L2 elements and the corresponding + infrastructure. + The v4l2videodec element replaces the mfcdec element. + ∘ rtpstreampay and rtpstreamdepay elements for transmitting + RTP packets over a stream API (e.g. TCP) according to + RFC 4571. + ∘ rtprtx elements for standard compliant implementation of + retransmissions, integrated into the rtpmanager plugin. + ∘ audiomixer element that mixes multiple audio streams together + into a single one while keeping synchronization. This is + planned to become the replacement of the adder element. + ∘ OpenNI2 plugin for 3D cameras like the Kinect camera. + ∘ OpenEXR plugin for decoding high-dynamic-range EXR images. + ∘ curlsshsink and curlsftpsink to write files via SSH/SFTP. + ∘ videosignal, ivfparse and sndfile plugins ported from 0.10. + ∘ avfvideosrc, vtdec and other elements were ported from 0.10 and + are available on OS X and iOS now. • Other changes: - ∘ gst-libav now uses libav 9 - ∘ Static linking of plugins is supported now (also in 1.0.7) - ∘ rtspsrc: add support for NetClientClock: when the server suggests a - GstNetTimeProvider in the SDP, set up a GstNetClientClock that - slaves to the remote clock and suggest this clock in provide_clock. - Simplifies synchronized playback of a resource from an RTSP server. - gst-rtsp-server now supports adding this to the SDP and can provide - a network clock - ∘ RTP retransmission / NACK support and big RTP jitterbuffer improvements - ∘ SRTP and DTLS support - ∘ Changes to many elements and core to use the correct sticky event - order and also not lose any important sticky events during flushing - ∘ >1000 fixed bug reports, and many other bug fixes and other - improvements everywhere that had no bug report + ∘ gst-libav now uses libav 10, and gained support for H265/HEVC. + ∘ Support for hardware codecs and special memory types has been + improved with bugfixes and feature additions in various plugins + and base classes. + ∘ Various bugfixes and improvements to buffering in queue2 and + multiqueue elements. + ∘ dvbsrc supports more delivery mechanisms and other features + now, including DVB S2 and T2 support. + ∘ The MPEGTS library has support for many more descriptors. + ∘ Major improvements to tsdemux, especially time related. + ∘ souphttpsrc now has support for keep-alive connections, + compression, configurable number of retries and configuration + for SSL certificate validation. + ∘ hlsdemux has undergone major refactoring and works more + reliable now and supports more HLS features like trick modes. + Also fragments are pushed downstream while they're downloaded + now instead of waiting for each fragment to finish. + ∘ videoflip can automatically flip based on the orientation tag. + ∘ openjpeg supports the OpenJPEG2 API. + ∘ gst-rtsp-server supports SRTP and MIKEY now. + ∘ Lots of fixes for coverity warnings all over the place. + ∘ 400+ fixed bug reports, and many other bug fixes and other + improvements everywhere that had no bug report. Things to look out for: - • Single header includes for all libraries, e.g. #include - <gst/video/video.h> - this was needed for some bindings. - • Stricter (correct) caps subset checking in some cases where this was - not correct before. Caps will now always fail to be a compatible - subset of another set of caps if the subset caps are missing some - fields that the superset caps have. This might lead to not-negotiated - errors if caps are incomplete now. However, it also prevents possible - data corruption caused by piping data formatted in an - incompatible/unexpected way into some elements. Check your h264 caps - for stream-format and alignment fields and AAC caps for the - stream-format field. This change will also be included in the next - stable 1.0.8 release. - • Stricter checking for missing events and correct sticky event order - (stream-start, caps, segment) in some places; this is not enabled in - stable releases by default, but you may get warnings when using git - builds, development releases or when compiling with - -UG_DISABLE_ASSERT in CFLAGS - • x264enc now outputs data in byte-stream by default if downstream has - ANY caps (e.g. appsink without caps set, filesink, udpsink, - tcpserversink etc.) - • The MPEG TS demuxer posts messages contain the PMT, PAT, etc. in a - different format now. This new format uses the data structures from - the new MPEGTS library - • The GstContext API has changed between 1.1.4 and 1.1.90 + • The eglglessink element was removed and replaced by the glimagesink + element. + • The mfcdec element was removed and replaced by v4l2videodec. + • osxvideosink is only available in OS X 10.6 or newer. @@ -1,16 +1,28 @@ +Release notes for GStreamer Base Plugins 1.3.1 -Release notes for GStreamer Base Plugins 1.2.0 +The GStreamer team is pleased to announce the first release of the unstable +1.3 release series. The 1.3 release series is adding new features on top of +the 1.0 and 1.2 series and is part of the API and ABI-stable 1.x release +series of the GStreamer multimedia framework. The unstable 1.3 release series +will lead to the stable 1.4 release series in the next weeks, and newly added +API can still change until that point. -The GStreamer team is proud to announce a new feature release -in the 1.x stable series of the -core of the GStreamer streaming media framework. -The 1.x series is a stable series targeted at end users. -It is not API or ABI compatible with the stable 0.10.x series. -It is, however, parallel installable with the 0.10.x series and -will not affect an existing 0.10.x installation. +Binaries for Android, iOS, Mac OS X and Windows will be provided separately +during the unstable 1.3 release series. + + + +The versioning scheme that is used in general is that 1.x.y is API and +ABI backwards compatible with previous 1.x.y releases. If x is an even +number it is a stable release series and all releases in this series +will only contain important bugfixes, e.g. the 1.0 series with 1.0.7. If +x is odd it is a development release series that will lead to the next +stable release series 1.x+1 and contains new features and bigger +changes. During the development release series, new API can still +change. @@ -54,16 +66,80 @@ contains a set of less supported plugins that haven't passed the gst-libav contains a set of codecs plugins based on libav (formerly gst-ffmpeg) - - - - Bugs fixed in this release - * 708667 : rtspconnection: leaks file descriptors/child sources - * 708372 : dmabuf: sys/mman.h: No such file or directory - * 708590 : adder: Should send its segment before checking for eos - * 708606 : video-frame: offsets are not copied from metadata + * 684030 : typefinding: mp4 with video and dts ES detected as DTS audio + * 725078 : audiobasesink: clip start samples to match clipped timestamp from skew algorithm + * 708633 : adder: Should not take channel mask in consideration when in mono or stereo + * 540941 : v4l2: RGB32 should be mapped to xRGB instead of RGBx + * 646577 : rtppayload: Make RTP time information accessible + * 670690 : audioresample: missing configure checks for SSE / SSE2 + * 678402 : Device discovery/listing replacement for GstPropertyProbe + * 678590 : subparse: Add support for LRC subtitles + * 679031 : playbin/playsink: Add support for audio and video filters + * 687183 : videodecoder: Allow to negotiate a buffer pool before output format is known + * 702230 : audioringbuffer: Don't access timestamps array if not acquired + * 707361 : video: Add support for 64x32 tiled NV12 color format + * 707636 : dashdemux: offline playback not buffering correctly + * 708680 : typefind: Add typefind function for H265 + * 708921 : pbutils: Add codec-utility functions to support h265 + * 708991 : audiocdsrc: invalid musicbrainz discids because of trailing data tracks + * 709588 : encodebin: Handle changes in encoding_profile::restriction during playback + * 709646 : videotestsrc: Could implement duration query when num-buffers is set + * 709755 : alsa: add channel map API support + * 709814 : [examples/overlay] avoid to unref sink if not found. Also fix logic to find a sink in one of the example. + * 709858 : theoraenc: Do nothing when flushing the encoder when no caps were set + * 710760 : videoconvert: remove unneeded guint comparison + * 711094 : videodecoder: improve max-error handling + * 711258 : sdp: fix duplicate 'const' declaration warnings + * 712798 : videometa: add GstVideoGLTextureUploadMeta buffer pool option + * 719383 : rtpbasepayload: Perfect timestamps confusingly explained + * 719415 : rtpbasepayload: Expose running time of last processed buffer + * 719850 : convertframe: remove trivial memory leak + * 719890 : videodecoder: Add API to get the currently pending, parsed frame size + * 720103 : videodecoder: Introduce sink_query/src_query + * 720124 : tests/examples/overlay/qt-videooverlay.cpp has incorrect include from Qt + * 720162 : tests: Add test for rtpbasepayload/-depayload + * 720205 : playback: add video/x-raw(ANY) to default raw caps + * 720215 : sdp: parse encryption key field + * 720219 : rtsptransport: allow getting mime type by profile + * 720389 : videodecoder: should release buffer pool sooner + * 720810 : audio/video: Initialize all {audio|video}info fields + * 720999 : Missing annotation for GstColorBalance interface + * 721103 : test-effect-switch errors out with not-negotiated after a while + * 721701 : videoconvert: I420 to BGRA conversion is slower than in 0.10 + * 721953 : pango: basetextoverlay: handle video/x-raw(ANY) if downstream supports the GstVideoOverlayCompositionMeta API + * 722330 : streamsplitter: negotiation problems with parsers + * 722491 : playbin: remove duplicate assignment + * 722682 : oggmux: problems with vp8 stream + * 723096 : decodebin: Make it possible to register multiple handlers to decodebin's autoplug-select signal + * 723271 : videotestsrc: fix a warning if downstream does not propose a buffer pool + * 723328 : gstrtpbase(|de)payload: add more unit tests and fix bugs + * 723492 : gst-plugins-base: Do not build check tests for disabled plugins + * 723507 : jsseek: Add missing HAVE_X check + * 724393 : rtspconnection: allow specifying an anchor certificate database + * 724509 : audioconvert: outputs silence when converting certain mono caps to certain other mono caps + * 724828 : playbin: improve autoplug_query_caps return + * 724893 : playsinkconvertbin: improve gst_play_sink_convert_bin_getcaps return + * 725034 : all plugin sets but -base don't install gtk-doc docs without '--enable-gtk-doc' + * 725206 : rtspconnection: Missing include file + * 725479 : gst-plugins-base: Ignore gcov intermediate files + * 725521 : docs: Fix argument and annotation typos, add missing annotations and remove duplicate section + * 725658 : Removing some GnomeVFS left bits + * 725837 : pango: textoverlay: lot of warnings in debug log with framerate=0/1 + * 725878 : rtspconnection: headers in GET response not configurable for tunnels + * 725898 : Lose data when producing data faster than sendt during tunneling rtps/rtp(TCP) + * 726433 : rtspconnection: setsockopt() argument 4 is not properly casted for W32 + * 726641 : rtspconnection: connection_poll() not working correctly + * 727498 : videodecoder: deactivates downstream bufferpool + * 728772 : rtspconnection: stuck in teardown + * 728845 : gst-play: add option to supply input media-files from a playlist file + * 728907 : rtspconnection: add more tests + * 729114 : audiodecoder: default caps nego will manually fixate non-mutable caps + * 729117 : rtpbuffer: fix memory leak when gst_rtp_buffer_map fails + * 729195 : videotestsrc: undefined behaviour in left-shift + * 729321 : playbin/subtitleoverlay: Deadlock when changing subtitle track while PAUSED + * 704933 : uridecodebin: allow progressive buffering with more media types ==== Download ==== @@ -100,10 +176,63 @@ subscribe to the gstreamer-devel list. Contributors to this release + * Adrien Schwartzentruber + * Aleix Conchillo Flaque + * Aleix Conchillo Flaqué + * Alessandro Decina + * Andres Gomez + * Antoine Jacoutot + * Antonio Ospite + * Arun Raghavan + * Bastien Nocera + * Christian Fredrik Kalager Schaller + * David Svensson Fors * Edward Hervey + * Eric Trousset + * George Kiagiadakis + * Göran Jönsson + * Haakon Sporsheim + * Hans Månsson + * Holger Kaelberer + * Jan Schmidt + * Jihyun Cho + * Johannes Dewender + * John Bassett + * Josep Torra + * Julien Isorce + * Justin Joy + * Lionel Landwerlin + * Luis de Bethencourt + * Mark Nauwelaerts + * Matej Knopp * Mathieu Duponchelle + * MathieuDuponchelle + * Matthew Waters + * Matthieu Bouron + * Nicola Murino + * Nicolas Dufresne * Ognyan Tonchev + * Olivier Crête + * Rafał Mużyło + * Ravi Kiran K N + * Reynaldo H. Verdejo Pinochet * Sebastian Dröge + * Sebastian Rasmussen + * Sjoerd Simons + * Sreerenj Balachandran + * Stefan Sauer + * Stephan Sundermann + * Stian Selnes + * Stéphane Cerveau + * Takashi Iwai + * Thiago Santos + * Thibault Saunier * Tim-Philipp Müller + * Todd Agulnick + * Tom Greenwood + * Vincent Penquerc'h + * William Grant * Wim Taymans -
\ No newline at end of file + * Wonchul Lee + * Руслан Ижбулатов + diff --git a/configure.ac b/configure.ac index dc2d33f84..8ea5adbc1 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],[1.3.0.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base]) +AC_INIT([GStreamer Base Plug-ins],[1.3.1],[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, 300, 0, 300) +AS_LIBTOOL(GST, 301, 0, 301) dnl *** required versions of GStreamer stuff *** -GST_REQ=1.3.0.1 +GST_REQ=1.3.1 dnl *** autotools stuff **** diff --git a/docs/plugins/gst-plugins-base-plugins.args b/docs/plugins/gst-plugins-base-plugins.args index f0ff83b03..af2ff6aa3 100644 --- a/docs/plugins/gst-plugins-base-plugins.args +++ b/docs/plugins/gst-plugins-base-plugins.args @@ -339,6 +339,26 @@ </ARG> <ARG> +<NAME>GstPlaySink::audio-filter</NAME> +<TYPE>GstElement*</TYPE> +<RANGE></RANGE> +<FLAGS>rw</FLAGS> +<NICK>Audio filter</NICK> +<BLURB>the audio filter(s) to apply, if possible.</BLURB> +<DEFAULT></DEFAULT> +</ARG> + +<ARG> +<NAME>GstPlaySink::video-filter</NAME> +<TYPE>GstElement*</TYPE> +<RANGE></RANGE> +<FLAGS>rw</FLAGS> +<NICK>Video filter</NICK> +<BLURB>the video filter(s) to apply, if possible.</BLURB> +<DEFAULT></DEFAULT> +</ARG> + +<ARG> <NAME>GstPlayBin::audio-sink</NAME> <TYPE>GstElement*</TYPE> <RANGE></RANGE> @@ -639,6 +659,26 @@ </ARG> <ARG> +<NAME>GstPlayBin::audio-filter</NAME> +<TYPE>GstElement*</TYPE> +<RANGE></RANGE> +<FLAGS>rw</FLAGS> +<NICK>Audio filter</NICK> +<BLURB>the audio filter(s) to apply, if possible.</BLURB> +<DEFAULT></DEFAULT> +</ARG> + +<ARG> +<NAME>GstPlayBin::video-filter</NAME> +<TYPE>GstElement*</TYPE> +<RANGE></RANGE> +<FLAGS>rw</FLAGS> +<NICK>Video filter</NICK> +<BLURB>the video filter(s) to apply, if possible.</BLURB> +<DEFAULT></DEFAULT> +</ARG> + +<ARG> <NAME>GstDecodeBin::caps</NAME> <TYPE>GstCaps*</TYPE> <RANGE></RANGE> diff --git a/docs/plugins/gst-plugins-base-plugins.hierarchy b/docs/plugins/gst-plugins-base-plugins.hierarchy index a3eee22f2..32d2ae15b 100644 --- a/docs/plugins/gst-plugins-base-plugins.hierarchy +++ b/docs/plugins/gst-plugins-base-plugins.hierarchy @@ -108,6 +108,7 @@ GObject GstPadTemplate GstPlugin GstPluginFeature + GstDeviceMonitorFactory GstElementFactory GstTypeFindFactory GstRegistry diff --git a/docs/plugins/inspect/plugin-adder.xml b/docs/plugins/inspect/plugin-adder.xml index 295245652..9592d1cc2 100644 --- a/docs/plugins/inspect/plugin-adder.xml +++ b/docs/plugins/inspect/plugin-adder.xml @@ -3,10 +3,10 @@ <description>Adds multiple streams</description> <filename>../../gst/adder/.libs/libgstadder.so</filename> <basename>libgstadder.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-alsa.xml b/docs/plugins/inspect/plugin-alsa.xml index b80da1f2b..4306b21ff 100644 --- a/docs/plugins/inspect/plugin-alsa.xml +++ b/docs/plugins/inspect/plugin-alsa.xml @@ -3,10 +3,10 @@ <description>ALSA plugin library</description> <filename>../../ext/alsa/.libs/libgstalsa.so</filename> <basename>libgstalsa.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-app.xml b/docs/plugins/inspect/plugin-app.xml index fcd46bfb4..7303dadff 100644 --- a/docs/plugins/inspect/plugin-app.xml +++ b/docs/plugins/inspect/plugin-app.xml @@ -3,10 +3,10 @@ <description>Elements used to communicate with applications</description> <filename>../../gst/app/.libs/libgstapp.so</filename> <basename>libgstapp.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-audioconvert.xml b/docs/plugins/inspect/plugin-audioconvert.xml index 3887a4b84..65db16821 100644 --- a/docs/plugins/inspect/plugin-audioconvert.xml +++ b/docs/plugins/inspect/plugin-audioconvert.xml @@ -3,10 +3,10 @@ <description>Convert audio to different formats</description> <filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename> <basename>libgstaudioconvert.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-audiorate.xml b/docs/plugins/inspect/plugin-audiorate.xml index 78fcbc6fc..665771492 100644 --- a/docs/plugins/inspect/plugin-audiorate.xml +++ b/docs/plugins/inspect/plugin-audiorate.xml @@ -3,10 +3,10 @@ <description>Adjusts audio frames</description> <filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename> <basename>libgstaudiorate.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-audioresample.xml b/docs/plugins/inspect/plugin-audioresample.xml index 96e9f68fb..96c3df31d 100644 --- a/docs/plugins/inspect/plugin-audioresample.xml +++ b/docs/plugins/inspect/plugin-audioresample.xml @@ -3,10 +3,10 @@ <description>Resamples audio</description> <filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename> <basename>libgstaudioresample.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-audiotestsrc.xml b/docs/plugins/inspect/plugin-audiotestsrc.xml index 172fb5ddb..a4e08b756 100644 --- a/docs/plugins/inspect/plugin-audiotestsrc.xml +++ b/docs/plugins/inspect/plugin-audiotestsrc.xml @@ -3,10 +3,10 @@ <description>Creates audio test signals of given frequency and volume</description> <filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename> <basename>libgstaudiotestsrc.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-cdparanoia.xml b/docs/plugins/inspect/plugin-cdparanoia.xml index 24312709f..e2f6da976 100644 --- a/docs/plugins/inspect/plugin-cdparanoia.xml +++ b/docs/plugins/inspect/plugin-cdparanoia.xml @@ -3,10 +3,10 @@ <description>Read audio from CD in paranoid mode</description> <filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename> <basename>libgstcdparanoia.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-encoding.xml b/docs/plugins/inspect/plugin-encoding.xml index 25615f85f..f74c7e3d6 100644 --- a/docs/plugins/inspect/plugin-encoding.xml +++ b/docs/plugins/inspect/plugin-encoding.xml @@ -3,10 +3,10 @@ <description>various encoding-related elements</description> <filename>../../gst/encoding/.libs/libgstencodebin.so</filename> <basename>libgstencodebin.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-gio.xml b/docs/plugins/inspect/plugin-gio.xml index fe7f87062..9b3f79404 100644 --- a/docs/plugins/inspect/plugin-gio.xml +++ b/docs/plugins/inspect/plugin-gio.xml @@ -3,10 +3,10 @@ <description>GIO elements</description> <filename>../../gst/gio/.libs/libgstgio.so</filename> <basename>libgstgio.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-ivorbisdec.xml b/docs/plugins/inspect/plugin-ivorbisdec.xml index 72645e8b6..bb337d215 100644 --- a/docs/plugins/inspect/plugin-ivorbisdec.xml +++ b/docs/plugins/inspect/plugin-ivorbisdec.xml @@ -3,7 +3,7 @@ <description>Vorbis Tremor decoder</description> <filename>../../ext/vorbis/.libs/libgstivorbisdec.so</filename> <basename>libgstivorbisdec.so</basename> - <version>1.2.0</version> + <version>1.3.1</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 6231cd995..1db9fa7b9 100644 --- a/docs/plugins/inspect/plugin-libvisual.xml +++ b/docs/plugins/inspect/plugin-libvisual.xml @@ -3,10 +3,10 @@ <description>libvisual visualization plugins</description> <filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename> <basename>libgstlibvisual.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-ogg.xml b/docs/plugins/inspect/plugin-ogg.xml index 29049d904..69acc2476 100644 --- a/docs/plugins/inspect/plugin-ogg.xml +++ b/docs/plugins/inspect/plugin-ogg.xml @@ -3,10 +3,10 @@ <description>ogg stream manipulation (info about ogg: http://xiph.org)</description> <filename>../../ext/ogg/.libs/libgstogg.so</filename> <basename>libgstogg.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-pango.xml b/docs/plugins/inspect/plugin-pango.xml index 2eaa4e07e..648b77ec8 100644 --- a/docs/plugins/inspect/plugin-pango.xml +++ b/docs/plugins/inspect/plugin-pango.xml @@ -3,10 +3,10 @@ <description>Pango-based text rendering and overlay</description> <filename>../../ext/pango/.libs/libgstpango.so</filename> <basename>libgstpango.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-playback.xml b/docs/plugins/inspect/plugin-playback.xml index 197c9d865..20ca92438 100644 --- a/docs/plugins/inspect/plugin-playback.xml +++ b/docs/plugins/inspect/plugin-playback.xml @@ -3,10 +3,10 @@ <description>various playback elements</description> <filename>../../gst/playback/.libs/libgstplayback.so</filename> <basename>libgstplayback.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-subparse.xml b/docs/plugins/inspect/plugin-subparse.xml index c3a056518..85dce2298 100644 --- a/docs/plugins/inspect/plugin-subparse.xml +++ b/docs/plugins/inspect/plugin-subparse.xml @@ -3,10 +3,10 @@ <description>Subtitle parsing</description> <filename>../../gst/subparse/.libs/libgstsubparse.so</filename> <basename>libgstsubparse.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-tcp.xml b/docs/plugins/inspect/plugin-tcp.xml index 78556e260..404e7c413 100644 --- a/docs/plugins/inspect/plugin-tcp.xml +++ b/docs/plugins/inspect/plugin-tcp.xml @@ -3,10 +3,10 @@ <description>transfer data over the network via TCP</description> <filename>../../gst/tcp/.libs/libgsttcp.so</filename> <basename>libgsttcp.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-theora.xml b/docs/plugins/inspect/plugin-theora.xml index 02d9711f7..9015d28fd 100644 --- a/docs/plugins/inspect/plugin-theora.xml +++ b/docs/plugins/inspect/plugin-theora.xml @@ -3,10 +3,10 @@ <description>Theora plugin library</description> <filename>../../ext/theora/.libs/libgsttheora.so</filename> <basename>libgsttheora.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-typefindfunctions.xml b/docs/plugins/inspect/plugin-typefindfunctions.xml index bbcdb91c4..8290951c0 100644 --- a/docs/plugins/inspect/plugin-typefindfunctions.xml +++ b/docs/plugins/inspect/plugin-typefindfunctions.xml @@ -3,10 +3,10 @@ <description>default typefind functions</description> <filename>../../gst/typefind/.libs/libgsttypefindfunctions.so</filename> <basename>libgsttypefindfunctions.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> </elements> diff --git a/docs/plugins/inspect/plugin-videoconvert.xml b/docs/plugins/inspect/plugin-videoconvert.xml index 9e6983b64..c37d73454 100644 --- a/docs/plugins/inspect/plugin-videoconvert.xml +++ b/docs/plugins/inspect/plugin-videoconvert.xml @@ -3,10 +3,10 @@ <description>Colorspace conversion</description> <filename>../../gst/videoconvert/.libs/libgstvideoconvert.so</filename> <basename>libgstvideoconvert.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-videorate.xml b/docs/plugins/inspect/plugin-videorate.xml index 483e7d85a..6b1df2ed1 100644 --- a/docs/plugins/inspect/plugin-videorate.xml +++ b/docs/plugins/inspect/plugin-videorate.xml @@ -3,10 +3,10 @@ <description>Adjusts video frames</description> <filename>../../gst/videorate/.libs/libgstvideorate.so</filename> <basename>libgstvideorate.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-videoscale.xml b/docs/plugins/inspect/plugin-videoscale.xml index 96f56c267..597cc0411 100644 --- a/docs/plugins/inspect/plugin-videoscale.xml +++ b/docs/plugins/inspect/plugin-videoscale.xml @@ -3,10 +3,10 @@ <description>Resizes video</description> <filename>../../gst/videoscale/.libs/libgstvideoscale.so</filename> <basename>libgstvideoscale.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-videotestsrc.xml b/docs/plugins/inspect/plugin-videotestsrc.xml index 1fe3dd911..442e5b3fd 100644 --- a/docs/plugins/inspect/plugin-videotestsrc.xml +++ b/docs/plugins/inspect/plugin-videotestsrc.xml @@ -3,10 +3,10 @@ <description>Creates a test video stream</description> <filename>../../gst/videotestsrc/.libs/libgstvideotestsrc.so</filename> <basename>libgstvideotestsrc.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-volume.xml b/docs/plugins/inspect/plugin-volume.xml index ae3c44fd0..d904e878a 100644 --- a/docs/plugins/inspect/plugin-volume.xml +++ b/docs/plugins/inspect/plugin-volume.xml @@ -3,10 +3,10 @@ <description>plugin for controlling audio volume</description> <filename>../../gst/volume/.libs/libgstvolume.so</filename> <basename>libgstvolume.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-vorbis.xml b/docs/plugins/inspect/plugin-vorbis.xml index 3255c277c..e4ba41f14 100644 --- a/docs/plugins/inspect/plugin-vorbis.xml +++ b/docs/plugins/inspect/plugin-vorbis.xml @@ -3,10 +3,10 @@ <description>Vorbis plugin library</description> <filename>../../ext/vorbis/.libs/libgstvorbis.so</filename> <basename>libgstvorbis.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-ximagesink.xml b/docs/plugins/inspect/plugin-ximagesink.xml index d7136d085..77554cf62 100644 --- a/docs/plugins/inspect/plugin-ximagesink.xml +++ b/docs/plugins/inspect/plugin-ximagesink.xml @@ -3,10 +3,10 @@ <description>X11 video output element based on standard Xlib calls</description> <filename>../../sys/ximage/.libs/libgstximagesink.so</filename> <basename>libgstximagesink.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-xvimagesink.xml b/docs/plugins/inspect/plugin-xvimagesink.xml index 331c2280d..24f88be28 100644 --- a/docs/plugins/inspect/plugin-xvimagesink.xml +++ b/docs/plugins/inspect/plugin-xvimagesink.xml @@ -3,10 +3,10 @@ <description>XFree86 video output plugin using Xv extension</description> <filename>../../sys/xvimage/.libs/libgstxvimagesink.so</filename> <basename>libgstxvimagesink.so</basename> - <version>1.3.0.1</version> + <version>1.3.1</version> <license>LGPL</license> <source>gst-plugins-base</source> - <package>GStreamer Base Plug-ins git</package> + <package>GStreamer Base Plug-ins source release</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/gst-libs/gst/audio/gstaudiopack-dist.c b/gst-libs/gst/audio/gstaudiopack-dist.c index 2a9a02553..004e20321 100644 --- a/gst-libs/gst/audio/gstaudiopack-dist.c +++ b/gst-libs/gst/audio/gstaudiopack-dist.c @@ -195,8 +195,8 @@ void audio_orc_splat_u64 (guint64 * ORC_RESTRICT d1, int p1, int n); #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX) #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX) #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX) -#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8)) -#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24)) +#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8)) +#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24)) #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56)) #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset))) #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff)) diff --git a/gst-libs/gst/video/video-orc-dist.c b/gst-libs/gst/video/video-orc-dist.c index 89833224c..fad5d6aa1 100644 --- a/gst-libs/gst/video/video-orc-dist.c +++ b/gst-libs/gst/video/video-orc-dist.c @@ -193,8 +193,8 @@ void video_orc_merge_linear_u8 (orc_uint8 * ORC_RESTRICT d1, #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX) #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX) #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX) -#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8)) -#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24)) +#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8)) +#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24)) #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56)) #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset))) #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff)) diff --git a/gst-plugins-base.doap b/gst-plugins-base.doap index 6b3dc5605..8ab9386dd 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>1.3.1</revision> + <branch>1.3</branch> + <name></name> + <created>2014-05-03</created> + <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.3.1.tar.xz" /> + </Version> + </release> + + <release> + <Version> <revision>1.2.0</revision> <branch>1.2</branch> <name></name> diff --git a/gst/adder/gstadderorc-dist.c b/gst/adder/gstadderorc-dist.c index d0077a11c..7d6e50211 100644 --- a/gst/adder/gstadderorc-dist.c +++ b/gst/adder/gstadderorc-dist.c @@ -160,8 +160,8 @@ void adder_orc_add_volume_f64 (double *ORC_RESTRICT d1, #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX) #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX) #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX) -#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8)) -#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24)) +#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8)) +#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24)) #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56)) #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset))) #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff)) diff --git a/gst/audioconvert/gstaudioconvertorc-dist.c b/gst/audioconvert/gstaudioconvertorc-dist.c index 46fdfface..ee3a539d9 100644 --- a/gst/audioconvert/gstaudioconvertorc-dist.c +++ b/gst/audioconvert/gstaudioconvertorc-dist.c @@ -228,8 +228,8 @@ void audio_convert_orc_pack_double_s32_swap (guint8 * ORC_RESTRICT d1, #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX) #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX) #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX) -#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8)) -#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24)) +#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8)) +#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24)) #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56)) #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset))) #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff)) diff --git a/gst/videoconvert/gstvideoconvertorc-dist.c b/gst/videoconvert/gstvideoconvertorc-dist.c index cd3291144..362f5273a 100644 --- a/gst/videoconvert/gstvideoconvertorc-dist.c +++ b/gst/videoconvert/gstvideoconvertorc-dist.c @@ -237,8 +237,8 @@ void video_convert_orc_convert_I420_BGRA (guint8 * ORC_RESTRICT d1, #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX) #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX) #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX) -#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8)) -#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24)) +#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8)) +#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24)) #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56)) #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset))) #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff)) @@ -8582,18 +8582,14 @@ video_convert_orc_convert_I420_BGRA (guint8 * ORC_RESTRICT d1, var54 = var42 - var43; /* 3: splatbw */ var55.i = ((var54 & 0xff) << 8) | (var54 & 0xff); - /* 4: loadupib */ - var56 = - (i & 1) ? ((orc_uint8) ptr5[i >> 1] + (orc_uint8) ptr5[(i >> 1) + 1] + - 1) >> 1 : ptr5[i >> 1]; + /* 4: loadupdb */ + var56 = ptr5[i >> 1]; /* 6: subb */ var57 = var56 - var44; /* 7: splatbw */ var58.i = ((var57 & 0xff) << 8) | (var57 & 0xff); - /* 8: loadupib */ - var59 = - (i & 1) ? ((orc_uint8) ptr6[i >> 1] + (orc_uint8) ptr6[(i >> 1) + 1] + - 1) >> 1 : ptr6[i >> 1]; + /* 8: loadupdb */ + var59 = ptr6[i >> 1]; /* 10: subb */ var60 = var59 - var45; /* 11: splatbw */ @@ -8756,18 +8752,14 @@ _backup_video_convert_orc_convert_I420_BGRA (OrcExecutor * ORC_RESTRICT ex) var54 = var42 - var43; /* 3: splatbw */ var55.i = ((var54 & 0xff) << 8) | (var54 & 0xff); - /* 4: loadupib */ - var56 = - (i & 1) ? ((orc_uint8) ptr5[i >> 1] + (orc_uint8) ptr5[(i >> 1) + 1] + - 1) >> 1 : ptr5[i >> 1]; + /* 4: loadupdb */ + var56 = ptr5[i >> 1]; /* 6: subb */ var57 = var56 - var44; /* 7: splatbw */ var58.i = ((var57 & 0xff) << 8) | (var57 & 0xff); - /* 8: loadupib */ - var59 = - (i & 1) ? ((orc_uint8) ptr6[i >> 1] + (orc_uint8) ptr6[(i >> 1) + 1] + - 1) >> 1 : ptr6[i >> 1]; + /* 8: loadupdb */ + var59 = ptr6[i >> 1]; /* 10: subb */ var60 = var59 - var45; /* 11: splatbw */ @@ -8852,7 +8844,7 @@ video_convert_orc_convert_I420_BGRA (guint8 * ORC_RESTRICT d1, 1, 1, 14, 1, 128, 0, 0, 0, 14, 4, 127, 0, 0, 0, 16, 2, 16, 2, 16, 2, 16, 2, 16, 2, 20, 2, 20, 2, 20, 2, 20, 2, 20, 2, 20, 2, 20, 1, 20, 1, 20, 1, 20, 4, 65, 38, 4, 16, - 151, 32, 38, 46, 38, 5, 65, 38, 38, 16, 151, 33, 38, 46, 38, 6, + 151, 32, 38, 45, 38, 5, 65, 38, 38, 16, 151, 33, 38, 45, 38, 6, 65, 38, 38, 16, 151, 34, 38, 90, 32, 32, 24, 90, 35, 34, 25, 71, 35, 32, 35, 90, 37, 33, 26, 71, 37, 32, 37, 90, 36, 33, 27, 71, 36, 32, 36, 90, 32, 34, 28, 71, 36, 36, 32, 159, 38, 35, 159, 39, @@ -8893,13 +8885,13 @@ video_convert_orc_convert_I420_BGRA (guint8 * ORC_RESTRICT d1, ORC_VAR_D1); orc_program_append_2 (p, "splatbw", 0, ORC_VAR_T1, ORC_VAR_T7, ORC_VAR_D1, ORC_VAR_D1); - orc_program_append_2 (p, "loadupib", 0, ORC_VAR_T7, ORC_VAR_S2, + orc_program_append_2 (p, "loadupdb", 0, ORC_VAR_T7, ORC_VAR_S2, ORC_VAR_D1, ORC_VAR_D1); orc_program_append_2 (p, "subb", 0, ORC_VAR_T7, ORC_VAR_T7, ORC_VAR_C1, ORC_VAR_D1); orc_program_append_2 (p, "splatbw", 0, ORC_VAR_T2, ORC_VAR_T7, ORC_VAR_D1, ORC_VAR_D1); - orc_program_append_2 (p, "loadupib", 0, ORC_VAR_T7, ORC_VAR_S3, + orc_program_append_2 (p, "loadupdb", 0, ORC_VAR_T7, ORC_VAR_S3, ORC_VAR_D1, ORC_VAR_D1); orc_program_append_2 (p, "subb", 0, ORC_VAR_T7, ORC_VAR_T7, ORC_VAR_C1, ORC_VAR_D1); diff --git a/gst/videoscale/gstvideoscaleorc-dist.c b/gst/videoscale/gstvideoscaleorc-dist.c index 0ecc1fa6b..85b37fd3d 100644 --- a/gst/videoscale/gstvideoscaleorc-dist.c +++ b/gst/videoscale/gstvideoscaleorc-dist.c @@ -153,8 +153,8 @@ void video_scale_orc_merge_bicubic_u8 (guint8 * ORC_RESTRICT d1, #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX) #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX) #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX) -#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8)) -#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24)) +#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8)) +#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24)) #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56)) #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset))) #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff)) diff --git a/gst/videotestsrc/gstvideotestsrcorc-dist.c b/gst/videotestsrc/gstvideotestsrcorc-dist.c index 63ca17a1e..c49a1ed19 100644 --- a/gst/videotestsrc/gstvideotestsrcorc-dist.c +++ b/gst/videotestsrc/gstvideotestsrcorc-dist.c @@ -121,8 +121,8 @@ void video_test_src_orc_splat_u32 (guint8 * ORC_RESTRICT d1, int p1, int n); #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX) #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX) #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX) -#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8)) -#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24)) +#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8)) +#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24)) #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56)) #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset))) #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff)) diff --git a/gst/volume/gstvolumeorc-dist.c b/gst/volume/gstvolumeorc-dist.c index 1a3f5195f..4d37493c1 100644 --- a/gst/volume/gstvolumeorc-dist.c +++ b/gst/volume/gstvolumeorc-dist.c @@ -148,8 +148,8 @@ void volume_orc_process_controlled_int8_2ch (gint8 * ORC_RESTRICT d1, #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX) #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX) #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX) -#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8)) -#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24)) +#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8)) +#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24)) #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56)) #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset))) #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff)) diff --git a/win32/common/_stdint.h b/win32/common/_stdint.h index 282caacf2..4bfdfcc23 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 1.2.0" -/* generated using gnu compiler gcc-4.8 (Debian 4.8.1-10) 4.8.1 */ +#define _GENERATED_STDINT_H "gst-plugins-base 1.3.1" +/* generated using gnu compiler Debian clang version 3.5-1 (trunk) (based on LLVM 3.5) */ #define _STDINT_HAVE_STDINT_H 1 #include <stdint.h> #endif diff --git a/win32/common/config.h b/win32/common/config.h index f64d65b13..8b655c99e 100644 --- a/win32/common/config.h +++ b/win32/common/config.h @@ -84,7 +84,7 @@ #define GST_PACKAGE_ORIGIN "Unknown package origin" /* GStreamer package release date/time for plugins as YYYY-MM-DD */ -#define GST_PACKAGE_RELEASE_DATETIME "2013-09-24" +#define GST_PACKAGE_RELEASE_DATETIME "2014-05-03" /* Define if static plugins should be built */ #undef GST_PLUGIN_BUILD_STATIC @@ -322,7 +322,7 @@ #define PACKAGE_NAME "GStreamer Base Plug-ins" /* Define to the full name and version of this package. */ -#define PACKAGE_STRING "GStreamer Base Plug-ins 1.2.0" +#define PACKAGE_STRING "GStreamer Base Plug-ins 1.3.1" /* Define to the one symbol short name of this package. */ #define PACKAGE_TARNAME "gst-plugins-base" @@ -331,7 +331,7 @@ #undef PACKAGE_URL /* Define to the version of this package. */ -#define PACKAGE_VERSION "1.2.0" +#define PACKAGE_VERSION "1.3.1" /* directory where plugins are located */ #ifdef _DEBUG @@ -365,7 +365,7 @@ #undef USE_TREMOLO /* Version number of package */ -#define VERSION "1.2.0" +#define VERSION "1.3.1" /* 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/gstrtsp-enumtypes.c b/win32/common/gstrtsp-enumtypes.c index ea8ea4af4..69ca5da87 100644 --- a/win32/common/gstrtsp-enumtypes.c +++ b/win32/common/gstrtsp-enumtypes.c @@ -281,6 +281,7 @@ gst_rtsp_header_field_get_type (void) "x-sessioncookie"}, {GST_RTSP_HDR_RTCP_INTERVAL, "GST_RTSP_HDR_RTCP_INTERVAL", "rtcp-interval"}, + {GST_RTSP_HDR_KEYMGMT, "GST_RTSP_HDR_KEYMGMT", "keymgmt"}, {GST_RTSP_HDR_LAST, "GST_RTSP_HDR_LAST", "last"}, {0, NULL, NULL} }; @@ -367,6 +368,8 @@ gst_rtsp_status_code_get_type (void) "unsupported-transport"}, {GST_RTSP_STS_DESTINATION_UNREACHABLE, "GST_RTSP_STS_DESTINATION_UNREACHABLE", "destination-unreachable"}, + {GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, + "GST_RTSP_STS_KEY_MANAGEMENT_FAILURE", "key-management-failure"}, {GST_RTSP_STS_INTERNAL_SERVER_ERROR, "GST_RTSP_STS_INTERNAL_SERVER_ERROR", "internal-server-error"}, {GST_RTSP_STS_NOT_IMPLEMENTED, "GST_RTSP_STS_NOT_IMPLEMENTED", diff --git a/win32/common/video-enumtypes.c b/win32/common/video-enumtypes.c index 9fd6acaed..a1f32112a 100644 --- a/win32/common/video-enumtypes.c +++ b/win32/common/video-enumtypes.c @@ -10,6 +10,7 @@ #include "colorbalance.h" #include "navigation.h" #include "video-chroma.h" +#include "video-tile.h" /* enumerations from "video-format.h" */ GType @@ -71,6 +72,8 @@ gst_video_format_get_type (void) {GST_VIDEO_FORMAT_GBR_10LE, "GST_VIDEO_FORMAT_GBR_10LE", "gbr-10le"}, {GST_VIDEO_FORMAT_NV16, "GST_VIDEO_FORMAT_NV16", "nv16"}, {GST_VIDEO_FORMAT_NV24, "GST_VIDEO_FORMAT_NV24", "nv24"}, + {GST_VIDEO_FORMAT_NV12_64Z32, "GST_VIDEO_FORMAT_NV12_64Z32", + "nv12-64z32"}, {0, NULL, NULL} }; GType g_define_type_id = g_enum_register_static ("GstVideoFormat", values); @@ -95,6 +98,7 @@ gst_video_format_flags_get_type (void) {GST_VIDEO_FORMAT_FLAG_COMPLEX, "GST_VIDEO_FORMAT_FLAG_COMPLEX", "complex"}, {GST_VIDEO_FORMAT_FLAG_UNPACK, "GST_VIDEO_FORMAT_FLAG_UNPACK", "unpack"}, + {GST_VIDEO_FORMAT_FLAG_TILED, "GST_VIDEO_FORMAT_FLAG_TILED", "tiled"}, {0, NULL, NULL} }; GType g_define_type_id = @@ -455,3 +459,38 @@ gst_video_chroma_flags_get_type (void) } return g_define_type_id__volatile; } + +/* enumerations from "video-tile.h" */ +GType +gst_video_tile_type_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GEnumValue values[] = { + {GST_VIDEO_TILE_TYPE_INDEXED, "GST_VIDEO_TILE_TYPE_INDEXED", "indexed"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_enum_register_static ("GstVideoTileType", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + +GType +gst_video_tile_mode_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GEnumValue values[] = { + {GST_VIDEO_TILE_MODE_UNKNOWN, "GST_VIDEO_TILE_MODE_UNKNOWN", "unknown"}, + {GST_VIDEO_TILE_MODE_ZFLIPZ_2X2, "GST_VIDEO_TILE_MODE_ZFLIPZ_2X2", + "zflipz-2x2"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_enum_register_static ("GstVideoTileMode", 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 177f19663..04f14783a 100644 --- a/win32/common/video-enumtypes.h +++ b/win32/common/video-enumtypes.h @@ -53,6 +53,12 @@ GType gst_video_chroma_method_get_type (void); #define GST_TYPE_VIDEO_CHROMA_METHOD (gst_video_chroma_method_get_type()) GType gst_video_chroma_flags_get_type (void); #define GST_TYPE_VIDEO_CHROMA_FLAGS (gst_video_chroma_flags_get_type()) + +/* enumerations from "video-tile.h" */ +GType gst_video_tile_type_get_type (void); +#define GST_TYPE_VIDEO_TILE_TYPE (gst_video_tile_type_get_type()) +GType gst_video_tile_mode_get_type (void); +#define GST_TYPE_VIDEO_TILE_MODE (gst_video_tile_mode_get_type()) G_END_DECLS #endif /* __GST_VIDEO_ENUM_TYPES_H__ */ |