summaryrefslogtreecommitdiff
path: root/ChangeLog
diff options
context:
space:
mode:
authorTim-Philipp Müller <tim@centricular.com>2019-09-23 11:09:38 +0100
committerTim-Philipp Müller <tim@centricular.com>2019-09-23 11:09:41 +0100
commitd7d290b64c1282398f3265522a33e093b7233310 (patch)
tree55e50ae4553a5d18d27ae29a42e82cf4298071ad /ChangeLog
parent1cc4f8ec24f4d02f456caf30ed3eb7f304eb192a (diff)
downloadgstreamer-plugins-good-d7d290b64c1282398f3265522a33e093b7233310.tar.gz
Release 1.16.11.16.1
Diffstat (limited to 'ChangeLog')
-rw-r--r--ChangeLog317
1 files changed, 317 insertions, 0 deletions
diff --git a/ChangeLog b/ChangeLog
index 628a07a4f..6fd5cfd4b 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,320 @@
+=== release 1.16.1 ===
+
+2019-09-23 11:09:38 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-plugins-good.doap:
+ * meson.build:
+ Release 1.16.1
+
+2019-09-23 11:09:38 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * 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-gtk.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-lame.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mpg123.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-qmlgl.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-shout2.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-twolame.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:
+ Update docs
+
+2019-09-23 11:09:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * po/fr.po:
+ * po/nb.po:
+ Update translations
+
+2019-09-08 20:43:17 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * gst/alpha/gstalpha.c:
+ alpha: Fix one_over_kc calculation
+ On arm/aarch64, converting from float directly to unsigned int uses
+ a different opcode and negative numbers result in 0. Cast to
+ signed int first.
+
+2019-08-07 18:29:25 -0400 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * tests/check/gst-plugins-good.supp:
+ valgrind: suppress Cond error coming from gnutls
+ taken from https://salsa.debian.org/debian/flatpak/commit/fb4a8dda211c4bc036781f2b0d706266e95ce068
+
+2019-06-04 13:39:00 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * tests/check/gst-plugins-good.supp:
+ supp: Ignore leaks caused by shout/sethostent
+ sethostent() seems to be using a global state and we endup with leaks from
+ that API when called through shout_init(). We had the option to only
+ ignore the shout case, but the impression is that if we have shout and
+ another sethostend user, as it's a global state, we may endup with a
+ different stack trace for the same leak. So in the end, we just ignore
+ memory allocated by sethostent in general.
+
+2019-08-22 00:18:51 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Fix incompatible type build warning
+ gstsouphttpsrc.c(2191): warning C4133:
+ '=': incompatible types - from 'guint (__cdecl *)(GType)' to 'GstURIType (__cdecl *)(GType)'
+
+2019-05-24 10:31:39 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: max-dropout-time gets cast to int32
+ So any value over MAXINT32 gets considered as negative and is silently ignored.
+
+2019-06-15 02:00:43 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: Clear clock master before unreffing
+ Make sure to clear any master clock on the media_clock
+ before unreffing it to release the timer callback that's
+ updating the clock and keeping it reffed.
+
+2019-08-01 15:02:23 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Use empty-array safe way to cleanup GPtrArray
+ Fix assertion fail
+ GLib-CRITICAL **: g_ptr_array_remove_range: assertion 'index_ < rarray->len' failed
+
+2019-08-06 22:27:40 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/ext/types-compat.h:
+ v4l2: Fix type compatibility issue with glibc 2.30
+ From now on, we will use linux/types.h on Linux, and use typedef of the
+ various flavour of BSD.
+ Fixes #635
+
+2019-07-31 21:55:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtpmanager/gstrtpfunnel.c:
+ rtpfunnel: forward correct segment when switching pad
+ Forwarding a single segment event from the pad that first gets
+ chained is incorrect: when that first event was sent by an element
+ such as x264enc, with its offset start, we end pushing out of segment
+ buffers for the other pad(s).
+ Instead, everytime the active pad changes, forward the appropriate
+ segment event.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1028
+
+2019-07-25 21:21:26 +0530 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * ext/gtk/gstgtkglsink.c:
+ * ext/gtk/gstgtkglsink.h:
+ gtkglsink: fix crash when widget is resized after element destruction
+ Prevent _size_changed_cb() to be called after gtkglsink has been finalized.
+ Fix #632
+
+2019-07-25 15:08:54 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: Don't dereference NULL input state if we have no caps in TIME segments
+ Simply assume that the JPEG frame is not going to be interlaced instead
+ of crashing.
+
+2019-07-22 10:28:50 +0200 Knut Andre Tidemann <knutandre.tidemann@zenitel.com>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtp: opuspay: fix memory leak in gst_rtp_opus_pay_setcaps.
+ The src caps were never dereferenced, causing a memory leak.
+
+2018-06-13 14:55:29 -0700 Song Bing <bing.song@nxp.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Fix drain() function return type
+ Return right type for drain() function.
+
+2019-05-21 15:25:03 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ * tests/check/elements/rtpssrcdemux.c:
+ rtpssrcdemux: Avoid taking streamlock out-of-band
+ In this change we now protect the internal srcpads list using the
+ stream lock and limit usage of the internal stream lock to
+ preventing data flowing on the other src pad type while creating
+ and signalling the new pad.
+ This fixes a deadlock with RTPBin shutdown lock. These two locks would
+ end up being taken in two different order, which caused a deadlock. More
+ generally, we should not rely on a streamlock when handling out-of-band
+ data, so as a side effect, we should not take a stream lock when
+ iterating internal links.
+
+2019-05-30 11:13:07 +0900 Damian Hobson-Garcia <dhobsong@igel.co.jp>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: return TRUE when buffer pool orphaning succeeds
+ When trying to orphan a buffer pool, successfully return and unref
+ the pool when the pool is either successfully stopped or orphaned.
+ Indicate failure and leave the pool untouched otherwise.
+
+2019-05-30 13:12:31 +0900 Damian Hobson-Garcia <dhobsong@igel.co.jp>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Free orphaned allocator resources when buffers are released
+ Allocator resources cannot be freed when a buffer pool is orphaned
+ while its buffers are in use. They should, however, be freed once those
+ buffers are no longer needed. This patch disposes of any buffers
+ belonging to an orphaned pool as they are released, and makes sure
+ that the allocator is cleaned up when the last buffer is returned.
+
+2019-05-27 18:08:54 +0900 Damian Hobson-Garcia <dhobsong@igel.co.jp>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Orphan buffer pool on object_stop if supported
+ Use V4L2 buffer orphaning, on recent kernels so that
+ the device can be restarted immediately with
+ a new buffer pool during renogatiation.
+
+2019-05-22 18:06:04 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Only set running time on finalizing sink element when in async-finalize mode
+ There is only a single sink element in async-finalize mode, and we would
+ keep the running time from previous fragments set in that case. As we
+ don't ever set the running time for the very last fragment on EOS, this
+ would mean that the closing time reported for the very last fragment is
+ the same as the closing time of the previous fragment.
+
+2019-05-14 17:36:14 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * tests/check/elements/rtpsession.c:
+ rtpsession: Always keep at least one NACK on early RTCP
+ We recently added code to remove outdate NACK to avoid using bandwidth
+ for packet that have no chance of arriving on time. Though, this had a
+ side effect, which is that it was to get an early RTCP packet with no
+ feedback into it. This was pretty useless but also had a side effect,
+ which is that the RTX RTT value would never be updated. So we we stared
+ having late RTX request due to high RTT, we'd never manage to recover.
+ This fixes the regression by making sure we keep at least one NACK in
+ this situation. This is really light on the bandwidth and allow for
+ quick recover after the RTT have spiked higher then the jitterbuffer
+ capacity.
+
+2019-04-24 13:47:54 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * tests/check/elements/rtpsession.c:
+ rtpsession: Call on-new-ssrc earlier
+ Right now, we may call on-new-ssrc after we have processed the first
+ RTP packet. This prevents properly configuring the source as some
+ property like "probation" are copied internally for use as a
+ decreasing counter. For this specific property, it prevents the
+ application from disabling probation on auxiliary sparse stream.
+ Probation is harmful on sparse streams since the probation algorithm
+ assume frequent and contiguous RTP packets.
+
+2019-04-24 13:54:12 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Add more information to probation warning
+
+2019-05-02 22:14:35 -0700 Thiago Santos <thiagossantos@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: do not try to send EOS with invalid seqnum
+ The second udpsrc (rtcp) might not have seen the segment event if it was
+ not enabled or if rtcp is not available on the server. So if the
+ application tries to send an EOS event it will try to set an invalid
+ seqnum to the event.
+
+2019-05-01 10:00:51 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ rtprawdepay: Don't get rid of the buffer pool on FLUSH_STOP
+ We expect there to be a pool as long as the caps are known and
+ FLUSH_STOP is not resetting the caps. Getting rid of the pool would
+ cause assertions.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/584
+
+2019-02-08 10:09:17 +0100 Danny Smith <dannys@axis.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Free storage when freeing session
+
+2019-04-23 10:10:01 +0100 Philippe Normand <philn@igalia.com>
+
+ * gst/audiofx/gstscaletempo.c:
+ scaletempo: Advertise interleaved layout in caps templates
+ Scaletempo doesn't support non-interleaved layout. Not explicitely stating this
+ would trigger critical warnings and a caps negotiation failure when scaletempo
+ is used as playbin audio-filter.
+ Patch suggested by George Kiagiadakis <george.kiagiadakis@collabora.com>.
+ Fixes #591
+
+2019-05-02 12:35:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitlab-ci.yml:
+ ci: use template from 1.16 branch
+
=== release 1.16.0 ===
2019-04-19 00:23:16 +0100 Tim-Philipp Müller <tim@centricular.com>