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 /ChangeLog | |
| parent | 876e28b9468a9f6fc9574fa9395e85599cf9e15a (diff) | |
Release 1.3.1
Diffstat (limited to 'ChangeLog')
| -rw-r--r-- | ChangeLog | 3437 |
1 files changed, 3435 insertions, 2 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 + + * gst-libs/gst/rtsp/gstrtspdefs.c: + * gst-libs/gst/rtsp/gstrtspdefs.h: + rtspdefs: remove outdated comments + +2014-05-02 15:09:35 +0100 Vincent Penquerc'h 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> |