gst-plugins-good 1.2.1
- From: Tim-Philipp Müller <install-module master gnome org>
- To: FTP Releases <ftp-release-list gnome org>
- Subject: gst-plugins-good 1.2.1
- Date: Thu, 21 Nov 2013 22:42:47 +0000 (UTC)
ChangeLog
=========
2013-11-09 Sebastian Dröge <slomo coaxion net>
* configure.ac:
releasing 1.2.1
2013-11-09 12:01:55 +0100 Sebastian Dröge <sebastian centricular com>
* po/de.po:
* po/id.po:
* po/sr.po:
po: Update translations
2013-11-08 17:59:24 +0100 Philippe Normand <philn igalia com>
* gst/wavenc/gstwavenc.c:
wavenc: generate a non-empty data header
Restore the behavior of the element to the state before commit
db29522a430e44450415ca3676abd1b77ee923d9. A non-empty header is
generated and when the EOS event is received the header is generated
again, this time with the correct size.
https://bugzilla.gnome.org/show_bug.cgi?id=711699
2013-10-07 14:27:21 -0300 Thiago Santos <ts santos partner samsung com>
* ext/soup/gstsouphttpsrc.c:
* ext/soup/gstsouphttpsrc.h:
souphttpsrc: do not emit EOS when connection drops
If the pipeline is stalled for too long, souphttpsrc will block and
stop fetching data from the network. This can cause the connection to
drop and souphttpsrc would handle it as an EOS. This patch makes it
persist and try to fetch more data until the end of the content length
or until receiving an error that it is beyong limits in case the content
is unknown.
https://bugzilla.gnome.org/show_bug.cgi?id=683536
2013-10-25 11:30:36 -0300 Thiago Santos <ts santos partner samsung com>
* gst/isomp4/qtdemux.c:
qtdemux: check if the end_time is defined before using it
Avoids sending EOS too soon because of overflow. Can happen on
fragmented mp4 playback.
2013-10-25 18:22:00 -0300 Thiago Santos <ts santos partner samsung com>
* gst/isomp4/qtdemux.c:
qtdemux: handle fragmented files with mdat before moofs
Assume a file with atoms in the following order: moov, mdat, moof,
mdat, moof ...
The first moov usually doesn't contain any sample entries atoms (or
they are all set to 0 length), because the real samples are signaled
at the moofs. In push mode, qtdemux parses the moov and then finds the mdat,
but then it has 0 entries and assumes it is EOS.
This patch makes it continue parsing in case it is a fragmented file so that
it might find the moofs and play the media.
https://bugzilla.gnome.org/show_bug.cgi?id=710623
2013-10-25 11:42:37 -0300 Thiago Santos <ts santos partner samsung com>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
qtdemux: When using a buffered mdat, store all received data for later use
In push mode, when qtdemux can't use a seek to skip the mdat buffer it has
to buffer it for later use.
The issue is that after parsing the next moov/moof, there might be some
trailing bytes from the next atom in the file. This data was being discarded
along with the already parsed moov/moof and playback would fail to continue
after the contents of this moov/moof are played.
This is particularly bad on fragmented files that have the mdat before the
corresponding moof. So you'd get:
mdat|moof|mdat|moof ...
When a moof was received, it usually came with some extra bytes that would
belong to the next mdat (because upstream doesn't care about atoms alignment).
So those bytes were being discarded and playback would fail.
This patch makes qtdemux store those extra bytes to reuse them later after the
mdat is emptied.
https://bugzilla.gnome.org/show_bug.cgi?id=710623
2013-11-07 09:49:55 +0100 Sebastian Dröge <sebastian centricular com>
* gst/udp/gstmultiudpsink.c:
multiudpsink: Also use the bind-port property if no bind-address was given
2013-11-07 00:51:12 +0100 Andoni Morales Alastruey <ylatuya gmail com>
* sys/osxaudio/gstosxcoreaudiohal.c:
osxaudiosink: fix segfault when we can't get the channels layout
2013-11-05 17:26:49 +0100 Sebastian Dröge <sebastian centricular com>
* gst/rtp/gstrtpvp8pay.c:
rtpvp8pay: Make Picture ID mode configurable and default to no picture ID
Some implementations (linphone) only support no picture at all in the
stream and will fail if one is provided.
https://bugzilla.gnome.org/show_bug.cgi?id=711497
2013-11-02 22:50:47 +0100 Rico Tzschichholz <ricotz ubuntu com>
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtspsrc.h:
rtsp: Add missing gio-2.0 deps and includes
2013-11-01 18:10:51 +0000 Olivier Crête <olivier crete collabora com>
* configure.ac:
Revert "configure: Require gst-plugins-base >= 1.2.1 for the TLS validation check flags in
GstRTSPConnection"
Version 1.2.1 doesn't exist yet, re-apply when it does
This reverts commit c98380985db3483ea78a8e738d544d1201d8ed1e.
2013-11-01 18:31:36 +0100 Sebastian Dröge <sebastian centricular com>
* gst/audiofx/audioiirfilter.c:
audioiirfilter: Fix initialization coefficient handling
Broke unit test.
2013-11-01 16:59:11 +0100 Sebastian Dröge <sebastian centricular com>
* configure.ac:
configure: Require gst-plugins-base >= 1.2.1 for the TLS validation check flags in GstRTSPConnection
2013-10-31 14:05:43 -0700 Aleix Conchillo Flaque <aleix oblong com>
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
rtspsrc: allow setting tls certificate validation flags
Added a new property "tls-validation-flags". If the url transport is
TLS, the validation flags will be set to the rtsp connection.
https://bugzilla.gnome.org/show_bug.cgi?id=711230
2013-10-31 22:43:49 +0100 Sebastian Dröge <sebastian centricular com>
* gst/audiofx/audiofxbaseiirfilter.c:
* gst/audiofx/audioiirfilter.c:
audioiirfilter: Don't crash if no filter coefficients are provided
...and by default use a identity filter.
https://bugzilla.gnome.org/show_bug.cgi?id=710215
2013-10-31 19:15:12 +0100 Sebastian Dröge <sebastian centricular com>
* ext/wavpack/gstwavpackenc.c:
wavpackenc: Fix writing of MD5 sums and other metadata blocks
These don't have the FINAL_BLOCK flag set.
2013-10-14 16:23:25 +0200 Ognyan Tonchev <ognyan axis com>
* gst/udp/gstmultiudpsink.c:
multiudpsink: Fix memory leak
Unmap all GstMemory of the current buffer when flushing.
https://bugzilla.gnome.org/show_bug.cgi?id=710110
2013-10-12 20:37:41 +0100 Tim-Philipp Müller <tim centricular net>
* gst/flv/gstflvmux.c:
flvmux: require stream-format=raw for mpeg-2 too, but don't require framed field
raw implies that it's framed already. Fixes .. ! faac ! flvmux
2013-10-10 13:52:35 +0200 Sebastian Dröge <slomo circular-chaos org>
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdec.h:
dvdec: Don't send segment event before caps
https://bugzilla.gnome.org/show_bug.cgi?id=709728
2013-10-09 17:46:33 +0200 Sebastian Dröge <slomo circular-chaos org>
* ext/dv/gstdvdemux.c:
dvdemux: Send stream-start, caps and segment events in the right order
https://bugzilla.gnome.org/show_bug.cgi?id=709728
2013-10-08 11:28:04 +0200 Sebastian Dröge <slomo circular-chaos org>
* gst/wavenc/gstwavenc.c:
wavenc: A-Law and Mu-Law don't have width/depth/signed caps fields
https://bugzilla.gnome.org/show_bug.cgi?id=709614
2013-10-07 12:54:11 +0200 Sebastian Dröge <slomo circular-chaos org>
* gst/deinterlace/tvtime/greedyh.c:
deinterlace: Fix handling of planar video formats in greedyh method
https://bugzilla.gnome.org/show_bug.cgi?id=709507
2013-10-04 13:34:09 +0200 Peter Korsgaard <peter korsgaard com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: O_CLOEXEC needs _GNU_SOURCE
On some systems (E.G. uClibc and older Glibc versions), O_CLOEXEC is only
defined when _GNU_SOURCE is specified, so do so.
_GNU_SOURCE needs to be defined before any system headers are included,
so move the fcntl.h section up.
https://bugzilla.gnome.org/show_bug.cgi?id=709423
2013-10-04 14:42:59 -0700 Reynaldo H. Verdejo Pinochet <r verdejo partner samsung com>
* gst/matroska/matroska-mux.c:
matroska: Do not write SegmentUID for WebM mux
WebM spec states SegmentUID is Unsupported. Files produced
with gstreamer without this change will spit an error like
this when passed to mkvalidator:
ERR201: Invalid 'SegmentUID' for profile 'webm' in Info at 192
2013-10-03 22:38:43 +0200 Mathieu Duponchelle <mduponchelle1 gmail com>
* gst/videomixer/videoconvert.c:
videomixer: Update videoconvert copy
https://bugzilla.gnome.org/show_bug.cgi?id=709390
2013-10-03 21:36:34 +0200 Mathieu Duponchelle <mduponchelle1 gmail com>
* gst/videomixer/videomixer2.c:
videomixer: Check if the pad needs reconfiguration in collected
https://bugzilla.gnome.org/show_bug.cgi?id=709384
2013-10-03 11:59:25 +0200 Sebastian Dröge <slomo circular-chaos org>
* gst/isomp4/qtdemux.c:
qtdemux: Add support for the mp2v fourcc for MPEG-2 video
https://bugzilla.gnome.org/show_bug.cgi?id=709270
2013-10-04 12:11:56 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: fix race in flush-start/flush-stop
When flush-stop arrives before we process the result of the _push() in the
loop function, we might pause even though we are not flushing anymore. Fix this
race by waiting for the srcpad loop function to completely pause after doing the
flush-start.
2013-10-03 14:39:35 +0100 Matthieu Bouron <matthieu bouron collabora com>
* ext/jpeg/gstjpegdec.c:
jpegdec: Relax sink caps
Since jpegdec already parse the jpeg stream, the sink caps could be
relaxed. This will allow jpegdec to be selected in more case and in
particular when the jpeg typefinder does not find the width and height.
https://bugzilla.gnome.org/show_bug.cgi?id=709352
2013-10-02 15:56:53 +0200 Ognyan Tonchev <ognyan axis com>
* gst/matroska/matroska-demux.c:
matroskademux: Fix memory leak
https://bugzilla.gnome.org/show_bug.cgi?id=709266
2013-09-30 12:24:32 +0200 Ognyan Tonchev <ognyan axis com>
* gst/rtp/gstrtpgstpay.c:
rtpgstpay: Fix memory leak
We were leaking the GList nodes of the pending buffers.
https://bugzilla.gnome.org/show_bug.cgi?id=709079
2013-09-30 12:31:00 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.h:
rtpjitterbuffer: fix race when updating the next_seqnum
If we were not waiting for the missing seqnum when we insert the lost packet
event in the jitterbuffer, we end up not updating the next_seqnum and wait
forever for the lost packets to arrive. Instead, keep track of the amount of
packets contained by the jitterbuffer item and update the next expected
seqnum only after pushing the buffer/event. This makes sure we correctly handle
GAPS in the sequence numbers.
2013-09-30 12:30:23 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: small debug improvement
2013-09-30 11:53:08 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/rtpjitterbuffer.c:
rtpjitterbuffer: reset skew does not reset clock-rate
Don't reset the clock-rate when we reset the skew correction algorithm.
Reset the skew correction algorithm when we change the clock-rate.
2013-09-30 11:16:32 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: pause timer when PAUSED
Also pause the timer when we go to the PAUSED state. It is possible that we
don't have a clock or base-time in PAUSED to perform the timeouts.
2013-09-30 11:15:25 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: improve debug
2013-09-27 15:05:04 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: also go into the loop function after connect
When we have opened the stream, go into the loop function so that we can
receive messages from the server.
2013-09-26 16:20:04 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/rtpjitterbuffer.c:
rtpjitterbuffer: don't calculate skew without rtptime
Skip trying to calculate the skew when we don't have an rtptime.
It causes problems when lost packet events are placed in the jitterbuffer.
2013-09-25 17:42:02 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: disable checks when linking pads
We know the pad links will work (and we don't check the return value
anyway).
2013-09-25 17:36:15 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpbin.c:
rtpbin: avoid some pad link checks
Link pads without checks, we know it will work.
2013-09-24 04:02:09 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: calculate some stats
2013-09-23 17:05:44 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: move send_lost_event function
Move the send_lost_event function to the do_lost_event handling, there is no
need to have a separate function.
2013-10-03 18:33:01 +0100 Tim-Philipp Müller <tim centricular net>
* sys/v4l2/gstv4l2object.c:
v4l2src: print probed caps as caps again in debug log
This got lost during refactoring.
2013-09-26 20:41:26 +0200 Hans Månsson <hansm axis com>
* gst/isomp4/gstqtmuxmap.c:
mp4mux: Do not require framerate in peer video caps
Remove the framerate restriction on the caps.
Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708864
2013-09-16 11:20:51 -0300 Thiago Santos <thiago sousa santos collabora com>
* gst/isomp4/qtdemux.c:
qtdemux: add code to parse creation time earlier than 1970
Use g_date_time seconds manipulation to allow to cover the quicktime
spec for creation_time. It uses seconds since 1904.
Both paths could be done using the generic approach of seconds since
1904 with GDateTime handling, but the first path using seconds from
1970 should be more commonly found and avoids a few objects creation and
ref/unref, so keep it there for performance.
Additionally, the code for handling seconds since 1970 changed from >
to >= because having 0 seconds since 1970 is also a valid case for that
path to handle.
https://bugzilla.gnome.org/show_bug.cgi?id=707975
2013-09-21 00:55:26 +0200 Matej Knopp <matej knopp gmail com>
* gst/matroska/matroska-demux.c:
matroskademux: update stream->pos when sending buffers so that gap events are not sent unnecessarily
https://bugzilla.gnome.org/show_bug.cgi?id=708505
2013-09-27 12:53:06 +0200 Matej Knopp <matej knopp gmail com>
* gst/matroska/matroska-demux.c:
matroskademux: move the check for subtitle buffer being null terminated before validating UTF-8
https://bugzilla.gnome.org/show_bug.cgi?id=707933
2013-09-25 12:55:21 +0200 Sebastian Dröge <slomo circular-chaos org>
* gst/isomp4/gstqtmux.c:
qtmux: Don't error out if downstream is not seekable for non-fragmented variants
Doing so would be a regression over 1.0 and breaks the unit test.
However the result will be most likely unusable, so let's post
a warning message on the bus.
2013-09-24 17:24:26 +0100 Tim-Philipp Müller <tim centricular net>
* README:
* common:
Automatic update of common submodule
From 6b03ba7 to 7412249
Download
========
https://download.gnome.org/sources/gst-plugins-good/1.2/gst-plugins-good-1.2.1.tar.xz (2.74M)
sha256sum: 660fa02dbe01086fcf702d87acc0ba5dde2559d6a11ecf438874afe504c50517
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