diff options
Diffstat (limited to 'ChangeLog')
| -rw-r--r-- | ChangeLog | 1255 |
1 files changed, 1253 insertions, 2 deletions
@@ -1,9 +1,1254 @@ +=== release 1.7.2 === + +2016-02-19 Sebastian Dröge <slomo@coaxion.net> + + * configure.ac: + releasing 1.7.2 + +2016-02-19 10:31:48 +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/mt.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/zh_HK.po: + * po/zh_TW.po: + po: Update translations + +2016-02-18 18:33:13 +0100 Philippe Normand <philn@igalia.com> + + * gst/isomp4/qtdemux.c: + qtdemux: plug leaks in cenc aux info parsing + +2016-02-18 13:43:07 +0000 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/Makefile.am: + tests: fix spurious souphttpsrc test timouts + Set GSETTINGS_BACKEND=memory, apparently there's something + about fork() and the dconf backend (or whatever else that + drags in or activates) that messes up locking and causes + timeouts due to deadlocks in g_mutex_lock(), since + everything works fine with CK_FORK=no as well. + +2016-02-18 11:10:14 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/matroska/matroska-demux.c: + matroskademux: Unmap wavpack header buffer after creating it + Otherwise it will be mapped writable all the time and we can't read from it + anywhere. + https://bugzilla.gnome.org/show_bug.cgi?id=762239 + +2015-12-08 18:49:40 +0100 Stian Selnes <stian@pexip.com> + + * tests/check/elements/rtpjitterbuffer.c: + rtpjitterbuffer: Add test for big seqnum gap handling + Make sure that the packets queued when detecting a big gap are pushed + after reset (5 consective seqnums) and not dropped. + https://bugzilla.gnome.org/show_bug.cgi?id=762211 + +2016-02-17 15:03:13 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtputils.h: + rtp: sprinkle some G_GNUC_INTERNAL for internal utils functions + +2016-02-09 13:17:00 +0000 Alex Ashley <bugzilla@ashley-family.net> + + * gst/isomp4/qtdemux.c: + qtdemux: only transform protected caps once + Commit 7873bede3134b15e5066e8d14e54d1f5054d2063 + (https://bugzilla.gnome.org/show_bug.cgi?id=760774) changed the + behaviour of qtdemux to call gst_qtdemux_configure_stream() for + every new moof. + When playing a protected stream, gst_qtdemux_configure_stream() + calls gst_qtdemux_configure_protected_caps(). The + gst_qtdemux_configure_protected_caps() function takes the original + media format, puts this in a field called "original-media-type" + and then changes the caps to "application/x-cenc". + The gst_qtdemux_configure_protected_caps() did not handle the case + of being called multiple times, causing it to incorrectly set the + caps. The second call was causing the caps to be set to: + application/x-cenc, original-media-type"application/x-cenc" + This commit makes gst_qtdemux_configure_protected_caps() check that + the caps have already been transformed, so that it only gets + changed once. + https://bugzilla.gnome.org/show_bug.cgi?id=761769 + +2016-02-17 13:26:02 +0000 Luis de Bethencourt <luisbg@osg.samsung.com> + + * gst/rtp/gstrtph264depay.c: + * gst/rtp/gstrtph265depay.c: + * gst/rtp/gstrtputils.c: + * gst/rtp/gstrtputils.h: + rtp: h264/h265: avoid duplication of read_golomb() + There is no need to have two identical implementations of the read_golomb + function. + https://bugzilla.gnome.org/show_bug.cgi?id=761606 + +2016-02-17 14:37:44 +0100 Ognyan Tonchev <ognyan@axis.com> + + * gst/matroska/matroska-demux.c: + matroskademux: Simple implementation of TRICKMODE_KEY_UNITS + When the trickmode key-units flag is set on the segment, simply skip + any sample on a video stream that isn't a keyframe + https://bugzilla.gnome.org/show_bug.cgi?id=762185 + +2015-08-21 14:15:18 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/matroska/matroska-demux.c: + matroska-demux: send GAP events for lagging audio and video streams too + Send GAP events for non-subtitle streams too if they lag too much + behind, but use a higher threshold than for subtitles. + This helps with fixing prerolling with a file where one of the + audio streams only has data starting from 19s onwards. It's not + a complete fix yet, it also requires changes elsewhere, such as + in baseparse, to make sure caps are propagated. + https://bugzilla.gnome.org/show_bug.cgi?id=614460 + https://bugzilla.gnome.org/show_bug.cgi?id=753899 + +2015-12-23 19:54:13 +0100 Stian Selnes <stian@pexip.com> + + * gst/rtp/Makefile.am: + * gst/rtp/gstrtp.c: + * gst/rtp/gstrtpvp9depay.c: + * gst/rtp/gstrtpvp9depay.h: + * gst/rtp/gstrtpvp9pay.c: + * gst/rtp/gstrtpvp9pay.h: + rtpvp9pay: rtpvp9depay: Initial implementation of draft 01 + Quick and dirty implementation of an RTP payloader and depayloader + for VP9. In particalur it assumes no spatial or temporal layering, + non-flexible mode, and some other bits and pieces. + https://bugzilla.gnome.org/show_bug.cgi?id=754773 + +2016-02-16 09:02:30 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * gst/avi/gstavidemux.c: + avidemux: Fix string memory leak + codec_name is not being freed in all conditions leading to memory leak + https://bugzilla.gnome.org/show_bug.cgi?id=762117 + +2015-12-10 12:15:52 +0100 Miguel París Díaz <mparisdiaz@gmail.com> + + * gst/rtpmanager/gstrtpbin.c: + * gst/rtpmanager/gstrtpbin.h: + rtpbin: add "get-session" signal + This gets the GstRTPSession element, as compared to the RTPSession object + that is returned by get-internal-session. + https://bugzilla.gnome.org/show_bug.cgi?id=759293 + +2016-02-16 00:19:00 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/Makefile.am: + * gst/rtp/gstrtp.c: + rtp: h265: hook up move RTP H.265 payloader/depayloader to build + https://bugzilla.gnome.org/show_bug.cgi?id=761606 + +2016-02-16 00:14:27 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtph265depay.c: + * gst/rtp/gstrtph265depay.h: + * gst/rtp/gstrtph265pay.c: + rtp: h265: use common meta utility functions + https://bugzilla.gnome.org/show_bug.cgi?id=761606 + +2016-02-05 18:18:31 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtph265depay.h: + * gst/rtp/gstrtph265pay.h: + * gst/rtp/gstrtph265types.h: + rtp: h265: remove codecparser dependency from h265 payloader/depayloader + Looks like it just uses the NAL enums and nothing else from + the codecparsers, and that's the only reason it had to be + moved from -good to -bad when it was originally added. We + can probably keep those NAL enums up to date enough, so let's + remove the codecparser dependency so it can be moved back into + -good. + https://bugzilla.gnome.org/show_bug.cgi?id=761606 + +2016-02-16 00:24:58 +0000 Tim-Philipp Müller <tim@centricular.com> + + Merge branch 'plugin-move-rtp-h265' + Move RTP H.265 payloader/depayloader from -bad to -good. + https://bugzilla.gnome.org/show_bug.cgi?id=761606 + +2016-02-05 15:34:51 +0000 Luis de Bethencourt <luisbg@osg.samsung.com> + + * gst/rtp/gstrtph265depay.c: + * gst/rtp/gstrtph265depay.h: + gstrtph265depay: keep consistency with rtph264depay + Use gst_rtp_drop_meta() and the same function prototype for + gst_rtp_copy_meta() to keep consistency with the RTP elements in + gst-plugins-good + +2016-02-05 13:56:34 +0000 Luis de Bethencourt <luisbg@osg.samsung.com> + + * gst/rtp/gstrtph265depay.c: + rtph265depay: fix termination of access unit + Only consider the access unit complete when the next-occurring VCL NAL unit + has the first bit after its NAL unit header equal to 1. + +2016-01-15 16:10:02 +0000 Luis de Bethencourt <luisbg@osg.samsung.com> + + * gst/rtp/gstrtph265depay.c: + rtph265depay: fix unneeded sub-buffer creation + We create a sub-buffer just to copy over its metas and then throw it + away immediately, just use the original input buffer directly. + +2016-01-15 15:56:59 +0000 Luis de Bethencourt <luisbg@osg.samsung.com> + + * gst/rtp/gstrtph265pay.c: + rtph265pay: add "send VPS/SPS/PPS with every key frame" mode + It's not enough to have timeout or event based VPS/SPS/PPS information + sent in RTP packets. There are some scenarios when key frames may appear + more frequently than once a second, in which case the minimum timeout + for "config-interval" of 1 second for sending VPS/SPS/PPS isn't enough. + It might also be desirable in general to make sure the VPS/SPS/PPS is + available with every keyframe (packet loss aside), so receivers can + actually pick up decoding immediately from the first keyframe if + VPS/SPS/PPS is not signaled out of band. + This commit adds the possibility to send VPS/SPS/PPS with every key frame. + This mode can be enabled by setting "config-interval" property to -1. In + this case the payloader will add VPS, SPS and PPS before every key (IDR) + frame. + https://bugzilla.gnome.org/show_bug.cgi?id=757892 + +2016-01-15 15:19:41 +0000 Luis de Bethencourt <luisbg@osg.samsung.com> + + * gst/rtp/gstrtph265pay.c: + * gst/rtp/gstrtph265pay.h: + rtph265pay: change config-interval property type from uint to int + This way we can use -1 as special value, which is nicer than MAXUINT. + https://bugzilla.gnome.org/show_bug.cgi?id=757892 + +2015-08-15 16:22:20 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265depay.c: + rtph265depay: make sure we call handle_nal for each NAL + Call handle_nal for each NAL in the STAP-A RTP packet. This makes sure + we correctly extract the SPS and PPS. + https://bugzilla.gnome.org/show_bug.cgi?id=730999 + +2015-08-15 14:45:34 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265pay.c: + rtph265pay: Copy metadata in the payloader, but only the relevant ones + The payloader didn't copy anything so far, the depayloader copied every + possible meta. Let's make it consistent and just copy all metas without + tags or with only the video tag. + https://bugzilla.gnome.org/show_bug.cgi?id=751774 + +2015-08-15 11:41:40 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265pay.c: + rtph265pay: Use GST_WARNING_OBJECT() instead of GST_WARNING() + https://bugzilla.gnome.org/show_bug.cgi?id=753228 + +2015-08-15 11:30:36 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265pay.c: + rtph265pay: fix potential crash when shutting down + A race condition in the state change function may cause buffers to be + unreffed while they are still used by the streaming thread in + gst_rtp_h265_pay_send_vps_sps_pps() resulting in a crash. Chain up to the + parent class first in the state change function to make sure streaming + has stopped and only then free those buffers. + https://bugzilla.gnome.org/show_bug.cgi?id=741381 + +2015-08-14 15:08:08 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265pay.c: + rtph265pay: fix buffer leak when using SPS/PPS + Fixes a buffer leak that would occur if the pipeline was shutdown while a + SPS/PPS header was being created. + https://bugzilla.gnome.org/show_bug.cgi?id=741271 + +2015-08-14 11:49:51 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265depay.c: + * gst/rtp/gstrtph265depay.h: + rtph265depay: copy metadata in the depayloader, but only the relevant ones + The payloader didn't copy anything so far, the depayloader copied every + possible meta. Let's make it consistent and just copy all metas without + tags or with only the video tag. + https://bugzilla.gnome.org/show_bug.cgi?id=751774 + +2015-08-12 17:54:52 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265depay.c: + rtph265depay: checking if depay has sps/pps nals before insertion + Related to: https://bugzilla.gnome.org/show_bug.cgi?id=753430 + https://bugzilla.gnome.org/show_bug.cgi?id=753228 + +2015-08-12 17:22:42 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265depay.c: + rtph265depay: only update the srcpad caps if something else than the codec_data changed + h264parse and gstrtph264depay do the same, let's keep the behaviour + consistent. As we now include the codec_data inside the stream, this causes + less caps renegotiation. + https://bugzilla.gnome.org/show_bug.cgi?id=753228 + +2015-08-12 16:43:48 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265depay.c: + rtph265depay: PPS replaces old PPS if it has the same id + https://bugzilla.gnome.org/show_bug.cgi?id=753228 + +2015-08-12 16:11:00 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265depay.c: + rtph265depay: Insert SPS/PPS NALs into the stream + rtph264depay does the same and this fixes decoding of some streams with 32 + SPS (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), + but the field in the codec_data for the number of SPS or PPS is only 5 + (or 8) bit. As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere. + This looks like a mistake in the part of the spect about the codec_data. + +2015-08-12 15:49:50 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265depay.c: + rtph265depay: implement process_rtp_packet() vfunc + For more optimised RTP packet handling: means we don't need to map the + input buffer again but can just re-use the mapping the base class has + already done. + Based on: https://bugzilla.gnome.org/show_bug.cgi?id=750235 + https://bugzilla.gnome.org/show_bug.cgi?id=753228 + +2015-08-12 15:14:50 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265depay.c: + rtph265depay: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP() + Switching to GST_BUFFER_TIMESTAMP() to be consistent with other rtp code. + +2015-08-12 14:59:53 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265depay.c: + rtph265depay: prevent trying to get 0 bytes from adapter + This causes an assertion and would lead to getting a NULL instead + of a buffer. Without proper checking this would easily lead to a + segfault. + Related to rpth264depay: https://bugzilla.gnome.org/show_bug.cgi?id=737199 + +2015-07-29 17:29:28 +0100 Luis de Bethencourt <luis@debethencourt.com> + + * gst/rtp/gstrtph265pay.c: + rtp: remove dead assignment + Value set to ret will be overwritten at least once at the end of the while + loop, removing assignment. + +2015-04-24 16:48:23 +0100 Luis de Bethencourt <luis.bg@samsung.com> + + * gst/rtp/gstrtph265pay.c: + remove unused enum items PROP_LAST + This were probably added to the enums due to cargo cult programming and are + unused. + +2015-03-06 14:54:41 +0000 Luis de Bethencourt <luis.bg@samsung.com> + + * gst/rtp/gstrtph265depay.c: + rtp: donl_present variable unused + donl_present is not implemented, yet the value is set and checked a few times. + Cleaning this. + CID #1249687 + +2015-01-08 15:36:04 +0000 Luis de Bethencourt <luis.bg@samsung.com> + + * gst/rtp/gstrtph265pay.c: + rtp: value truncated too short creates dead code + type is truncated to 0-31 with "& 0x1f", but right after that it is checks if + the value is equivalent to GST_H265_NAL_VPS, GST_H265_NAL_SPS, and + GST_H265_NAL_PPS (which are 32, 33, and 34 respectively). Obviously, this will + never be True if the value is maximum 31 after the truncation. + The intention of the code was to truncate to 0-63. + +2015-01-08 15:27:44 +0000 Luis de Bethencourt <luis.bg@samsung.com> + + * gst/rtp/gstrtph265depay.c: + rtp: fix nal unit type check + After further investigation the previous commit is wrong. The code intended to + check if the type is 39 or the ranges 41-44 and 48-55. Just like gsth265parse.c + does. Type 40 would not be complete. + +2015-01-08 13:47:09 +0000 Luis de Bethencourt <luis.bg@samsung.com> + + * gst/rtp/gstrtph265depay.c: + rtp: fix dead code and check for impossible values + nal_type is the index for a GstH265NalUnitType enum. There are two types of dead + code here: + First, after checking if nal_type is >= 39 there are two OR conditionals that + check if the value is in ranges higher than that number, so if nal_type >= 39 + falls in the True branch those other conditions aren't checked and if it falls + in the False branch and they are checked, they will always also be False. They + are redundant. + Second, the enum has a range of 0 to 40. So the checks for ranges higher than 41 + should never be True. + Removing this redundant checks. + CID 1249684 + +2014-10-16 10:34:01 +0200 Thijs Vermeir <thijsvermeir@gmail.com> + + * gst/rtp/gstrtph265depay.c: + * gst/rtp/gstrtph265depay.h: + * gst/rtp/gstrtph265pay.c: + * gst/rtp/gstrtph265pay.h: + rtp: add h265 RTP payloader + depayloader + +2016-02-15 11:51:46 +0900 Vineeth TM <vineeth.tm@samsung.com> + + * tests/check/elements/rtpmux.c: + tests: rtpmux: Fix element memory leak + https://bugzilla.gnome.org/show_bug.cgi?id=762057 + +2016-02-12 20:57:29 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst/monoscope/monoscope.c: + monoscope: rework the scaling code + The running average was wrong and the resulting scaling factor was only held in + place using the CLAMP. In addtion we are now convering quickly to volume + changes. + FInally now with this change, we can change the resolution defines and + everythign adjusts. + +2016-01-28 17:00:55 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst/monoscope/convolve.c: + * gst/monoscope/monoscope.c: + * gst/monoscope/monoscope.h: + monoscope: use constants in the drawing code + Make all the drawing ops be based on the constants. This way we can change + the fixed size at least at compile time. + +2016-01-28 09:51:17 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst/monoscope/gstmonoscope.c: + monoscope: replace hardcoded values by constants + This at least establishes the relationship. + +2016-01-28 09:43:12 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst/monoscope/convolve.c: + * gst/monoscope/convolve.h: + * gst/monoscope/monoscope.c: + * gst/monoscope/monoscope.h: + monoscpe: make the convolver use dynamic memory + Replace all #defines with members and initialize the convolver with a parameter. + +2016-01-28 08:56:44 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst/monoscope/README: + monoscope: update README + We can already create multiple instances. + +2016-01-28 08:53:35 +0100 Stefan Sauer <ensonic@users.sf.net> + + * gst/monoscope/convolve.c: + * gst/monoscope/monoscope.c: + monoscope: code cleanup + Use constants more often. Cleanup comments and add more to explain how things + work. + +2016-02-08 23:41:32 +0000 Luis de Bethencourt <luisbg@osg.samsung.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: remove check for impossible condition + Commit bd27a1f30b4458f2edee53c76dd07fb35904b61d added a few error handling + memory management checks. These check srccaps to see if it needs to be + unreferenced before returning, in the case of invalid_caps this goto jump + always happens before srccaps is set, so it will always be NULL in this + error label. + CID #1352035 + +2016-02-08 12:48:46 +0100 Piotr Drąg <piotrdrag@gmail.com> + + * po/POTFILES.in: + po: update POTFILES + https://bugzilla.gnome.org/show_bug.cgi?id=761705 + +2016-02-08 15:31:55 +0000 Luis de Bethencourt <luisbg@osg.samsung.com> + + * sys/v4l2/gstv4l2allocator.c: + v4l2allocator: Fix spelling of reenqueueing + To match commit 7d7074cef0272cd5155098bfc2bda6849dd89267. I love the idea + of aiming for the maximum number of consecutive vowels. + +2016-02-08 10:17:49 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * sys/v4l2/gstv4l2allocator.c: + v4l2allocator: Fix spelling of queueing + Didn't know which one to choose between queuing and queueing, so I picked + the one with the biggest amount of vowels in a row ;-P (both are + acceptable apparently) + +2016-02-07 15:02:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * ext/jpeg/gstjpegdec.c: + jpegdec: Don't pass the same data over and over + We already pass the entire frame to the decoder. If the decoder ask for + more data, don't pass the same data again as this leads to infinit loop. + Instead, simply fail the fill function to signal the problem with that + frame. It will then be skipped properly. + https://bugzilla.gnome.org/show_bug.cgi?id=761670 + +2016-02-08 00:10:33 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/matroska/lzo.c: + matroska: get rid of _stdint.h include + +2016-02-05 20:00:57 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * tests/check/Makefile.am: + tests: extend the AM_TESTS_ENVIRONMENT from check.mak + To get the CK_DEFAULT_TIMEOUT defined for all tests + https://bugzilla.gnome.org/show_bug.cgi?id=761472 + +2016-02-05 18:04:31 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * autogen.sh: + * common: + Automatic update of common submodule + From 86e4663 to b64f03f + +2016-01-30 18:43:30 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtp/gstrtpjpegpay.c: + rtpjpegpay: Skip APP and JPG markers and print warnings for unknown markers + For APP/JPG markers the size is following and we have to skip that. This is + not really a problem unless the marker contains e.g. a preview JPEG or + something else that we might interprete as another marker. + +2016-01-26 22:37:30 +0900 Seungha Yang <sh.yang@lge.com> + + * gst/isomp4/qtdemux.c: + qtdemux: fix framerate calculation for fragmented format + qtdemux calculates framerate using duration and the number of sample. + In case of fragmented mp4 format, however, the number of sample can + be figure out after parsing every moof box. Because qtdemux does not + parse every moof in QTDEMUX_STATE_HEADER state, it will cause incorrect + framerate calculation. + This patch will triger gst_qtdemux_configure_stream() for every new moof. + Then, framerate will be calculated by using duration and n_samples of the moof. + https://bugzilla.gnome.org/show_bug.cgi?id=760774 + +2016-01-28 22:36:23 +0900 Seungha Yang <sh.yang@lge.com> + + * gst/isomp4/qtdemux.c: + qtdemux: handling zero segment-duration edit list + Based on document ISO_IEC_14496-12, edit list box can have + segment duration as zero. It does not imply that media_start equals to + media_stop. But, it just indicates a sample which should be presented + at the first. This patch derives segment duration using media_time + and duration of file. And set derived duration to segment-duration. + https://bugzilla.gnome.org/show_bug.cgi?id=760781 + +2016-01-28 21:36:54 +0900 Seungha Yang <sh.yang@lge.com> + + * gst/isomp4/qtdemux.c: + * gst/isomp4/qtdemux.h: + qtdemux: expose streams with first moof for fragmented format + In case of push mode, qtdemux expose streams after got moov box. + We can not guarantee that a moov box has sample data such as sample duration + and the number of sample in stbl box for fragmented format case. + So, if a moov has no sample data, streams will not be exposed until get the first moof. + https://bugzilla.gnome.org/show_bug.cgi?id=760779 + +2016-01-27 18:48:17 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Check for subset instead of non-empty intersection for ACCEPT_CAPS + +2016-01-27 18:44:23 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Unset RECONFIGURE flag on srcpad whenever we configure new caps + Prevents double-negotiation during startup and in some other cases. + +2016-01-27 16:43:22 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/deinterlace.c: + deinterlace: Add negotiation unit tests for all 4 modes + These now check the output caps based on the input caps and a following + capsfilter and make sure the caps are exactly as expected. + https://bugzilla.gnome.org/show_bug.cgi?id=760995 + https://bugzilla.gnome.org/show_bug.cgi?id=720388 + +2016-01-26 17:39:20 +0100 Vivia Nikolaidou <vivia@toolsonair.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Do passthrough in auto mode if downstream only supports interlaced + If the following conditions are met: + 1) upstream and downstream caps are compatible + 2) upstream is interlaced + 3) downstream doesn't support progressive mode + then deinterlace will just do passthrough instead of failing to link. + This is done with the following scenario in mind: + videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace + name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. ! + queue ! deinterlace name=dein_desktop ! autovideosink + In this case, dein_src will do the deinterlacing. However, + videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace + name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. ! + queue ! deinterlace name=dein_desktop ! autovideosink t. ! queue ! + "video/x-raw,interlace-mode=interleaved" ! fakesink + In this case, caps auto-negotiation will make dein_file and dein_desktop do + the deinterlacing, while dein_src will be passthrough. + https://bugzilla.gnome.org/show_bug.cgi?id=760995 + +2016-01-26 18:05:51 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/deinterlace/gstdeinterlace.c: + * gst/deinterlace/gstdeinterlace.h: + deinterlace: Add mode=auto-strict + In this mode we will passthrough all progressive caps but interlaced caps must be + caps where we actually support deinterlacing. + This is the only difference between auto and auto-strict, auto would + passthrough all unsupported interlaced caps. + https://bugzilla.gnome.org/show_bug.cgi?id=720388 + +2016-01-26 17:50:30 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Implement reconfiguration a bit better + And e.g. consider reconfiguration caused by RECONFIGURE events too. + https://bugzilla.gnome.org/show_bug.cgi?id=720388 + +2016-01-26 11:57:09 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Rewrite caps negotiation + Previously the result of the CAPS query and ACCEPT_CAPS depended on what kind + of caps were last set, and e.g. if we last had interlaced caps or not. That's + just broken. + Also previously the handling of non-sysmem caps features was rather random and + unusuable. + Now the behaviour is the following, depending on the mode property: + 1) mode=disabled + Completely do passthrough of everything + 2) mode=interlaced + Only accept formats we can actually deinterlace, and accept interlaced + and progressive content and always run the deinterlacer and output + progressive content + 3) mode=auto (i.e. playbin) + Accept all progressive formats as passthrough, accept all formats that we + can deinterlace ourselves (which we do then), but also accept everything + else for which we then just passthrough. In auto mode, deinterlacing is best + effort: If we can, we deinterlace, if we can't we just output interlaced + content. + https://bugzilla.gnome.org/show_bug.cgi?id=720388 + https://bugzilla.gnome.org/show_bug.cgi?id=760553 + +2016-01-26 11:34:40 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Remove unused, obsolete bufferalloc code + +2016-01-26 18:50:38 +0100 Matej Knopp <matej.knopp@gmail.com> + + * gst/matroska/matroska-mux.c: + matroskamux: use A_AAC instead of A_AAC/MPEGx/y + Some GoogleCast compatible devices ignore A_AAC/MPEGx/y tracks; Also according to http://wiki.multimedia.cx/index.php?title=Matroska A_AAC/MPEGx/y is obsolete + https://bugzilla.gnome.org/show_bug.cgi?id=761144 + +2016-01-25 17:21:24 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com> + + * gst/isomp4/qtdemux.c: + * gst/rtp/gstrtph261pay.c: + gst: Fix unintialized variable warnings + While cross-compiling with Linaro GCC 5.1-2015.08, it complained + about a couple unitialized variables. + This patch initializes them to zero. + https://bugzilla.gnome.org/show_bug.cgi?id=761094 + +2016-01-25 15:03:23 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com> + + * gst/multifile/gstsplitmuxpartreader.c: + splitmuxsrc: print potentially negative offset with a sign + +2016-01-21 17:41:55 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * sys/v4l2/gstv4l2object.c: + v4l2: Re-add colorimetry field for RGB formats + This time, check if it's an RGB format and sets the transformation + matrix to identity. The rest of the colorimetry information is + meaningfull and shall be kept. + https://bugzilla.gnome.org/show_bug.cgi?id=759624 + +2016-01-22 10:03:50 +0100 Wim Taymans <wtaymans@redhat.com> + + * sys/v4l2/gstv4l2object.c: + v4l2: fix sRGB colorspace definition + V4l2 can also use the sRGB colorspace for YUV formats and thus needs a + default matrix. + +2016-01-21 15:29:46 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/debugutils/gsttaginject.c: + taginject: fix sample pipeline in docs + https://bugzilla.gnome.org/show_bug.cgi?id=679571 + +2016-01-21 10:49:44 +0100 Wim Taymans <wtaymans@redhat.com> + + * sys/v4l2/gstv4l2object.c: + v4l2: Add adobe colorspace support + Use the new primaries and transfer function for Adobe RGB. + Explicitly list the colorimetry instead of using the default GStreamer + ones. The defaults for BT2020, for example, do not match. + Explicitly set the matrix of SRGB to RGB. + +2016-01-20 13:41:33 +0200 Sebastian Dröge <sebastian@centricular.com> + + * ext/vpx/gstvp8enc.c: + vp8enc: Ensure that we always have valid frame user data before using it + Otherwise we're going to dereference NULL pointers. + +2016-01-20 10:02:48 +0200 Sebastian Dröge <sebastian@centricular.com> + + * ext/vpx/gstvpxdec.c: + vpxdec: Unref frame in all code paths of handle_frame() + https://bugzilla.gnome.org/show_bug.cgi?id=760666 + +2016-01-19 22:49:20 +0100 Thibault Saunier <tsaunier@gnome.org> + + * ext/vpx/gstvpxenc.c: + vpxenc: Unref frame on ERROR + All code paths for handle_frame() must somehow take ownership of the frame, be + it by actually unreffing, forwarding the frame elsewhere or storing it for + later. + http://bugzilla.gnome.org/show_bug.cgi?id=760666 + +2016-01-20 18:20:43 +1100 Jan Schmidt <jan@centricular.com> + + * sys/v4l2/gstv4l2deviceprovider.c: + v4l2: Don't free props structure twice. + gst_v4l2_device_provider_probe_device() frees the passed props + structure, don't free it again in the caller. + +2016-01-19 15:15:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * sys/v4l2/gstv4l2object.c: + v4l2object: Cleanup uneeded return statement + +2016-01-19 15:14:59 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com> + + * sys/v4l2/gstv4l2object.c: + v4l2object: Don't set colorimetry for non YUV formats + Setting colormetry in caps for RGB have no meaning, but worst it + confuses the converters downstream. + https://bugzilla.gnome.org/show_bug.cgi?id=759624 + +2016-01-19 13:01:17 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtpchannels.c: + * gst/rtp/gstrtpchannels.h: + rtp: fix compiler warnings with gcc-6 + In file included from gstrtpL16depay.h:27:0, + from gstrtp.c:73: + gstrtpchannels.h:154:33: error: 'channel_orders' defined but not used [-Werror=unused-const-variable] + static const GstRTPChannelOrder channel_orders[] = + +2016-01-19 14:57:03 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/wavparse/gstwavparse.c: + wavparse: Don't play anything after the end of the data chunk even when seeking + Especially in push mode we would completely ignore the size of the data chunk + when not stop position is given for the seek. Instead make sure that the end + offset is at most the end of the data chunk if known. + Without this we would output anything after the data chunk, possibly causing + loud noises if the media file is followed by an INFO chunk or an ID3 tag. + +2016-01-19 14:55:57 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/wavparse/gstwavparse.c: + wavparse: Don't do calculations with -1 offsets when handling SEGMENT events + We use that to signal "infinity", taking the difference between that and some + other value is not going to give us any useful result for the end offsets of + segments. + +2016-01-18 11:30:45 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + * gst/rtpmanager/rtpjitterbuffer.c: + * gst/rtpmanager/rtpjitterbuffer.h: + Revert "WIP: rtpjitterbuffer: Add RFC7273 media clock handling" + This reverts commit 271501f6576de4d141e7c2f618e28b9e3b1e5b38. + It wasn't meant to be pushed yet as the commit message indicates. + +2016-01-12 14:01:21 -0800 Aleix Conchillo Flaqué <aconchillo@gmail.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: handle rtcp/srtcp caps properly when using interleaved data + We check the stream profile and use the proper RTCP caps: + application/x-srtcp if we are using a secure profile and + application/x-rtcp otherwise. + https://bugzilla.gnome.org/show_bug.cgi?id=760556 + +2016-01-05 16:15:16 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + * gst/rtpmanager/rtpjitterbuffer.c: + * gst/rtpmanager/rtpjitterbuffer.h: + WIP: rtpjitterbuffer: Add RFC7273 media clock handling + +2016-01-15 11:36:35 +0000 Thibault Saunier <tsaunier@gnome.org> + + * ext/vpx/gstvpxenc.c: + vp8enc: Return FLOW_ERROR when an error accures + FALSE would mean FLOW_OK + https://bugzilla.gnome.org/show_bug.cgi?id=760666 + +2016-01-15 03:57:45 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * sys/osxaudio/gstosxcoreaudiohal.c: + osxaudio: break as soon as the device is found + No need to loop further if there's no side-effects for it + +2016-01-15 03:56:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com> + + * sys/osxaudio/gstosxaudioringbuffer.c: + * sys/osxaudio/gstosxcoreaudiohal.c: + osxaudio: Fix error handling when selecting/opening devices + Post an element error when the CoreAudio device cannot be selected or opened. + Also ensure that we post a GST_ERROR with more detail. + +2016-01-13 23:40:20 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/wavparse/gstwavparse.c: + wavparse: When flushing on EOS, don't process more data than the "data" size + Even if we have more data queued up when flushing than the size of the data + chunk, don't process and output it. If the data size is known, this likely + contains another chunk (e.g. an INFO chunk) or things like ID3 tags. Just + outputting them as if they were data is going to cause unexpected behaviour + and unpleasant audio noises. + +2014-08-29 15:40:23 +0200 Antonio Ospite <ao2@ao2.it> + + * tests/check/pipelines/wavenc.c: + tests: fix a thinko in the wavenc example + The code is supposed to follow somehow what the comment above says, that + is to have one channel with a wave of freq 440 and the other channel + with a wave of freq 880, but an off by one error results in frequencies + of 0 and 440. + https://bugzilla.gnome.org/show_bug.cgi?id=735673 + +2014-08-29 15:07:58 +0200 Antonio Ospite <ao2@ao2.it> + + * gst/interleave/interleave.c: + interleave: Fix the example by setting channel-masks in the sink pads + The current example does not work, it fails with: + ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0: Internal data flow error. + gstwavparse.c(2178): gst_wavparse_loop (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0: + streaming task paused, reason not-negotiated (-4) + This is because negotiation with wavenc gets messed up by the missing + channel positions configuration. + The proper way to define the channel layout when using the interleave + element in code would be to set the channel-positions property, but + gst-launch-1.0 does not know how to deal with arrays; so the example + pipeline works around the issue by setting the channel-masks in the sink + pads. + Also fix a repetition in the deinterleave example description + https://bugzilla.gnome.org/show_bug.cgi?id=735673 + +2016-01-11 16:29:55 +0000 Tim Sheridan <tim.sheridan@imgtec.com> + + * gst/audioparsers/gstsbcparse.c: + sbcparse: Fix frame length calculation + SBC frame length calculation wasn't being rounded up to the nearest byte + (as specified in the A2DP 1.0 specification, section 12.9). This could + cause 'stereo' and 'joint stereo' mode SBC streams to have incorrectly + calculated frame lengths. + Incorrect frame length calculation causes frame coalescing to fail, as + subsequent frames in the stream aren't found in the expected locations. + https://bugzilla.gnome.org/show_bug.cgi?id=742446 + +2016-01-10 22:54:12 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: demote warning on wrong reserved value to fixme + We are likely just parsing a backward-compatible stream we + don't fully support. + +2016-01-08 16:27:05 -0300 Thiago Santos <thiagoss@osg.samsung.com> + + * gst/imagefreeze/gstimagefreeze.c: + imagefreeze: simplify caps selection + The downstream caps query with a filter alraedy gives us the possible + intersection so there is no need to check it again with downstream + if it is supported. Just try to set it directly. + +2016-01-07 20:42:41 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtph264depay.c: + rtph264depay: fix unnecessary sub-buffer creation + We create a sub-buffer just to copy over its metas and then + throw it away immediately, just use the original input buffer + directly. + +2016-01-07 20:38:27 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtpdvdepay.c: + rtpdvdepay: fix unnecessary sub-buffer creation + We create a sub-buffer just to copy over its metas and then + throw it away immediately, just use the original input buffer + directly. + +2016-01-07 20:34:05 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtpamrdepay.c: + rtpamrdepay: fix unnecessary sub-buffer creation + We create a sub-buffer just to copy over its metas and then + throw it away immediately, just use the original input buffer + directly. + +2016-01-07 20:27:29 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtp/gstrtpvrawdepay.c: + rtpvrawdepay: fix major memory leak and performance issue + We call gst_rtp_buffer_get_payload() which creates a sub-buffer + of each input buffer, just to copy over metas, and then leak it. + https://bugzilla.gnome.org/show_bug.cgi?id=760289 + +2016-01-08 15:32:47 +0200 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/elements/rganalysis.c: + rganalysis: Fix compiler warnings in the unit test + elements/rganalysis.c:919:66: error: shifting a negative signed value is undefined + [-Werror,-Wshift-negative-value] + push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -1 << 14, 0)); + ~~ ^ + elements/rganalysis.c:929:69: error: shifting a negative signed value is undefined + [-Werror,-Wshift-negative-value] + push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -1 << 14)); + ~~ ^ + elements/rganalysis.c:939:64: error: shifting a negative signed value is undefined + [-Werror,-Wshift-negative-value] + push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -1 << 14)); + ~~ ^ + +2016-01-05 18:13:06 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: don't map buffer multiple times when parsing + +2016-01-07 18:20:30 +0200 Steven Hoving <sh@bigbrother.nl> + + * gst/matroska/matroska-read-common.c: + matroska: Store subtitle stream count in the correct variable + And don't override the video stream count instead. + +2016-01-05 18:59:06 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/equalizer/gstiirequalizernbands.c: + equalizer: The child-proxy API is GObject based in 1.x + Not GstObject anymore. + +2015-05-21 17:41:12 +0200 Pablo Anton <pablo.anton@vodalys-labs.com> + + * sys/v4l2/gstv4l2transform.c: + v4l2-*: Configuring output pool correctly for using drivers min_buffer if present. + Signed-off-by: Pablo Anton <pablo.anton@vodalys-labs.com> + https://bugzilla.gnome.org/show_bug.cgi?id=755736 + +2015-12-31 15:46:31 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: add debug msg on CRC mismatch while validating frame header + +2015-12-31 16:00:49 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: drop unneeded braces at _parse_frame() exit + Additionally, drop redundant comment & line break + +2015-12-31 15:55:18 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: minor grammar correction + +2015-12-31 15:34:57 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: update URLs on pointers to online spec + +2015-12-31 14:40:15 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: make buffer DTS setting explicitly unconditional + We are setting it to PTS regardless of block_strategy + +2015-12-31 14:21:40 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: add actual invalid block type to warning + For someone that read the spec is clear the only *invalid* + data block type is 127. For the rest, its useful information. + Additionally. values 7-126 are currently reserved by the + spec so the situation might change in the future. + +2015-12-31 14:12:36 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: use shift instead of mask & comp + We are only interested on the first bit of the first + byte of the metadata block header to figure out whether + is marked as the last one. The shift makes it quite + clearer. + +2015-12-31 12:52:13 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: warn on wishful parsing of weird headers + If we get anything from 7 to 126 as type when parsing + a metadata block header, we are likely dealing with a + FLAC stream version we don't fully understand. Issue + a warning if so. + Document function assumptions regarding the passed-on + type while at this. + +2015-12-31 11:33:45 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: show meaningful info on frame CRC check + As CRCs are calculated for the comparition already, we + might as well (cheaply) inform the user how the numbers + differ if a missmatched pair is found. + While at it: + Rephrase candidate-frame message to make more sense + +2015-12-31 02:40:43 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: drop remaining trailing whitespace + +2015-12-31 02:15:06 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: drop superflous else clauses + +2015-12-31 01:09:51 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: factor out buffer time and offset resetting + Avoids multiple occurrences of the same resetting pattern + +2015-12-31 00:54:48 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: move block handling by type out of _parse_frame() + +2015-10-07 18:51:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com> + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: replace duplicated codes to call new base sdp apis + https://bugzilla.gnome.org/show_bug.cgi?id=745880 + +2015-12-30 12:16:56 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: drop redundant return statement on _header_is_valid() + Fix the rather vague error message while at it. + +2015-12-30 01:56:26 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: rework gst_flac_parse_frame_is_valid() + drop unnecessary nesting looking for end of frame + +2015-12-30 00:37:04 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com> + + * gst/audioparsers/gstflacparse.c: + flacparse: factor out context clearing routine + +2015-12-29 18:05:56 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/matroska/matroska-demux.c: + matroskademux: Guard against no codec data in prores caps creation + CID 1346532 + +2015-12-29 17:58:38 +0200 Sebastian Dröge <sebastian@centricular.com> + + * ext/vpx/gstvpxdec.c: + vpxdec: Initialize buffer variable to NULL + False positive but trivial to fix and possibly causing compiler warnings at + some point in the future too. + CID 1346535 + +2015-07-27 15:53:26 +0200 Wim Taymans <wtaymans@redhat.com> + + * sys/v4l2/gstv4l2deviceprovider.c: + v4l2deviceprovider: add properties to the device + Add properties to the device with exactly the same keys and sematics + as what pulseaudio uses as property keys. + Also handle the case when a device is probed manually and not through gudev. + https://bugzilla.gnome.org//show_bug.cgi?id=759780 + +2015-12-25 11:41:19 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/audiofx/gstscaletempo.c: + scaletempo: Free the various buffers in GstBaseTransform::stop() + Previously we leaked them completely, but as they're specific to the caps + freeing them in stop() instead of finalize() makes most sense. + +2015-12-24 15:28:06 +0100 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + Back to development + === release 1.7.1 === -2015-12-24 Sebastian Dröge <slomo@coaxion.net> +2015-12-24 14:16:21 +0100 Sebastian Dröge <sebastian@centricular.com> + * ChangeLog: + * NEWS: + * RELEASE: * configure.ac: - releasing 1.7.1 + * docs/plugins/gst-plugins-good-plugins.args: + * docs/plugins/inspect/plugin-1394.xml: + * docs/plugins/inspect/plugin-aasink.xml: + * docs/plugins/inspect/plugin-alaw.xml: + * docs/plugins/inspect/plugin-alpha.xml: + * docs/plugins/inspect/plugin-alphacolor.xml: + * docs/plugins/inspect/plugin-apetag.xml: + * docs/plugins/inspect/plugin-audiofx.xml: + * docs/plugins/inspect/plugin-audioparsers.xml: + * docs/plugins/inspect/plugin-auparse.xml: + * docs/plugins/inspect/plugin-autodetect.xml: + * docs/plugins/inspect/plugin-avi.xml: + * docs/plugins/inspect/plugin-cacasink.xml: + * docs/plugins/inspect/plugin-cairo.xml: + * docs/plugins/inspect/plugin-cutter.xml: + * docs/plugins/inspect/plugin-debug.xml: + * docs/plugins/inspect/plugin-deinterlace.xml: + * docs/plugins/inspect/plugin-dtmf.xml: + * docs/plugins/inspect/plugin-dv.xml: + * docs/plugins/inspect/plugin-effectv.xml: + * docs/plugins/inspect/plugin-equalizer.xml: + * docs/plugins/inspect/plugin-flac.xml: + * docs/plugins/inspect/plugin-flv.xml: + * docs/plugins/inspect/plugin-flxdec.xml: + * docs/plugins/inspect/plugin-gdkpixbuf.xml: + * docs/plugins/inspect/plugin-goom.xml: + * docs/plugins/inspect/plugin-goom2k1.xml: + * docs/plugins/inspect/plugin-icydemux.xml: + * docs/plugins/inspect/plugin-id3demux.xml: + * docs/plugins/inspect/plugin-imagefreeze.xml: + * docs/plugins/inspect/plugin-interleave.xml: + * docs/plugins/inspect/plugin-isomp4.xml: + * docs/plugins/inspect/plugin-jack.xml: + * docs/plugins/inspect/plugin-jpeg.xml: + * docs/plugins/inspect/plugin-level.xml: + * docs/plugins/inspect/plugin-matroska.xml: + * docs/plugins/inspect/plugin-mulaw.xml: + * docs/plugins/inspect/plugin-multifile.xml: + * docs/plugins/inspect/plugin-multipart.xml: + * docs/plugins/inspect/plugin-navigationtest.xml: + * docs/plugins/inspect/plugin-oss4.xml: + * docs/plugins/inspect/plugin-ossaudio.xml: + * docs/plugins/inspect/plugin-png.xml: + * docs/plugins/inspect/plugin-pulseaudio.xml: + * docs/plugins/inspect/plugin-replaygain.xml: + * docs/plugins/inspect/plugin-rtp.xml: + * docs/plugins/inspect/plugin-rtpmanager.xml: + * docs/plugins/inspect/plugin-rtsp.xml: + * docs/plugins/inspect/plugin-shapewipe.xml: + * docs/plugins/inspect/plugin-shout2send.xml: + * docs/plugins/inspect/plugin-smpte.xml: + * docs/plugins/inspect/plugin-soup.xml: + * docs/plugins/inspect/plugin-spectrum.xml: + * docs/plugins/inspect/plugin-speex.xml: + * docs/plugins/inspect/plugin-taglib.xml: + * docs/plugins/inspect/plugin-udp.xml: + * docs/plugins/inspect/plugin-video4linux2.xml: + * docs/plugins/inspect/plugin-videobox.xml: + * docs/plugins/inspect/plugin-videocrop.xml: + * docs/plugins/inspect/plugin-videofilter.xml: + * docs/plugins/inspect/plugin-videomixer.xml: + * docs/plugins/inspect/plugin-vpx.xml: + * docs/plugins/inspect/plugin-wavenc.xml: + * docs/plugins/inspect/plugin-wavpack.xml: + * docs/plugins/inspect/plugin-wavparse.xml: + * docs/plugins/inspect/plugin-ximagesrc.xml: + * docs/plugins/inspect/plugin-y4menc.xml: + * gst-plugins-good.doap: + * win32/common/config.h: + Release 1.7.1 + +2015-12-24 13:19:24 +0100 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/mt.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/zh_HK.po: + * po/zh_TW.po: + Update .po files 2015-12-24 12:22:32 +0100 Sebastian Dröge <sebastian@centricular.com> @@ -120289,3 +121534,9 @@ Original commit message from CVS: add some files +2001-12-17 18:37:01 +0000 Thomas Vander Stichele <thomas@apestaart.org> + + building up speed + Original commit message from CVS: + building up speed + |