gst-plugins-good 1.7.2
- From: Sebastian Dröge <install-module master gnome org>
- To: FTP Releases <ftp-release-list gnome org>
- Subject: gst-plugins-good 1.7.2
- Date: Fri, 19 Feb 2016 12:17:15 +0000 (UTC)
ChangeLog
=========
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
Download
========
https://download.gnome.org/sources/gst-plugins-good/1.7/gst-plugins-good-1.7.2.tar.xz (3.09M)
sha256sum: eed4f6d2f7293e6e360e826e6fdb04ccb96612debed73d80d9c104896d2f0fc1
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