gst-plugins-good 1.1.90
- From: Tim-Philipp Müller <install-module master gnome org>
- To: FTP Releases <ftp-release-list gnome org>
- Subject: gst-plugins-good 1.1.90
- Date: Fri, 20 Sep 2013 08:55:41 +0000 (UTC)
ChangeLog
=========
2013-09-19 Sebastian Dröge <sebastian droege collabora co uk>
* configure.ac:
releasing 1.1.90
2013-09-19 09:45:18 +0200 Sebastian Dröge <slomo circular-chaos org>
* po/cs.po:
* po/nl.po:
* po/pl.po:
* po/uk.po:
* po/vi.po:
po: Update translations
2013-09-11 14:27:02 -0400 Olivier Crête <olivier crete collabora com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: dmabuf is not a singleton anymore
https://bugzilla.gnome.org/show_bug.cgi?id=707793
2013-09-16 13:53:45 -0300 Thiago Santos <thiago sousa santos collabora com>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: do not do http requests in READY
HEAD requests to discover if the server is seekable shouldn't be done in
READY as it might lock the main thread that is doing the state change.
https://bugzilla.gnome.org/show_bug.cgi?id=705371
2013-09-18 16:32:28 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: reevaluate the current timer after timeout
When we trigger the timeout logic of a timer, reevaluate it because it is
possible that it still has the lowest timeout.
2013-09-18 16:31:26 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: don't update time when unscheduled
Don't try to estimate the current time when we got unscheduled.
2013-09-18 16:29:37 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: init packet spacing on first buffer
Already init the packet spacing variables on the first buffer so that we can
calculate the spacing on the second buffer already.
2013-09-18 15:08:45 +0200 Wim Taymans <wim taymans collabora co uk>
* tests/check/elements/rtpjitterbuffer.c:
tests: fix comments
2013-09-18 14:57:09 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: push the lost event from the timer thread
Instead of pushing the lost event from the chain function, schedule a timeout
that will push the lost event from the timer thread. This avoid blocking the
upstream thread while we push and sync the event.
2013-09-18 14:23:55 +0200 Wim Taymans <wim taymans collabora co uk>
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: add another test
The test is modified slightly because the late lost packets are only
generated now when a large gap is received.
2013-09-18 14:12:47 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: round gap duration to multiple of duration
Make sure the gap duration in the lost event is a multiple of the packet
duration.
Enable another test.
2013-09-18 12:29:38 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/Makefile.am:
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: keep track of duration
Keep track of the estimated duration of missing packets and use it in the lost
event.
Enable another unit test
2013-09-18 11:59:28 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: handle large gaps with one lost event
When we have a large number of missing packets, generate one lost event for all
the packets that have no chance of being pushed out in time.
Fix and activate unit test for large gaps.
2013-09-18 11:56:38 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: refactor lost event sending
Also make sure we only increment the expected seqnum and last
output timestamp.
2013-09-17 23:21:09 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: refactor timeout triggers
2013-09-17 23:03:45 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: simplify the timeout code
Keep track of the current time in the timeout loop.
Loop over all timers and trigger all the expired ones, we can do this in the
same loop that selects the new best timer.
2013-09-17 23:01:17 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: rearrange timer update code
Also update the timers when retransmission is disabled. We need to
do this because when we added LOST timers when we detected missing packets and
we need to remove those timers when the packet finally arrives.
2013-09-17 22:02:04 +0100 Tim-Philipp Müller <tim centricular net>
* gst/videomixer/Makefile.am:
videomixer: link to libm for maths stuff
Fixes undefined references to rint and pow on ubuntu
build bot.
2013-09-17 15:19:42 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: release lock on shutdown
2013-09-17 15:11:41 +0200 Wim Taymans <wim taymans collabora co uk>
* tests/check/Makefile.am:
check: change for videomixer renamed orc file
2013-09-14 16:03:20 +0200 Matej Knopp <matej knopp gmail com>
* gst/isomp4/gstqtmux.c:
qtmux: remove MAX_TOLERATED_LATENESS
https://bugzilla.gnome.org/show_bug.cgi?id=707411
2013-09-16 15:54:37 +0200 Wim Taymans <wim taymans collabora co uk>
* tests/examples/rtp/client-H264-rtx.sh:
examples: we don't need the queue anymore
2013-09-16 15:53:47 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: use separate thread for timeouts
Use a separate thread for scheduling the timeouts instead of using the
downstream streaming thread that might block at any time.
2013-09-14 15:56:04 +0200 Matej Knopp <matej knopp gmail com>
* gst/isomp4/gstqtmux.c:
qtmux: set first_ts to DTS for streams that have DTS
https://bugzilla.gnome.org/show_bug.cgi?id=707340
2013-09-14 15:55:22 +0200 Matej Knopp <matej knopp gmail com>
* gst/isomp4/gstqtmux.c:
qtmux: make sure duration is a valid number for last buffer
https://bugzilla.gnome.org/show_bug.cgi?id=707340
2013-09-14 15:54:29 +0200 Matej Knopp <matej knopp gmail com>
* gst/isomp4/gstqtmux.c:
qtmux: use segment.start or last buffer end time in case of missing DTS
https://bugzilla.gnome.org/show_bug.cgi?id=707340
2013-09-03 18:14:04 +0200 Matej Knopp <matej knopp gmail com>
* gst/isomp4/gstqtmux.c:
Revert qtmux: Use buffer PTS if DTS is not set"
This reverts commit f72c3cf71fde622067f41f31a53978ba4c94469d.
https://bugzilla.gnome.org/show_bug.cgi?id=707340
2013-09-16 11:03:06 +0200 Sebastian Dröge <slomo circular-chaos org>
* gst/videomixer/videomixerorc-dist.c:
* gst/videomixer/videomixerorc-dist.h:
videomixer: Update orc generated files
https://bugzilla.gnome.org/show_bug.cgi?id=708131
2013-09-13 16:25:49 +0200 Olivier Crête <olivier crete collabora com>
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
rtpsession: Demux RTCP buffers from the RTP stream
If there are RTCP buffers in the RTP stream, process them as
RTCP. This way, we want receive streams following RFC 5761
https://bugzilla.gnome.org/show_bug.cgi?id=687657
2013-09-13 23:26:21 +1000 Jan Schmidt <thaytan noraisin net>
* gst/rtp/gstrtpL24depay.c:
rtp: Remove bogus extra caps from L24 template.
The extra caps entry in the template was making it sometimes
get plugged for any dynamically allocated payload type.
2013-09-13 12:40:41 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
rtpbin: use PacketInfo for the sender
Avoid mapping the packet multiple times when sending RTP.
2013-09-13 12:22:36 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
rtpbin: store more in the PacketInfo
Store all info in the PacketInfo so that we can avoid mapping the packet
multiple times.
2013-09-13 11:32:52 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpstats.h:
session: store more in the PacketInfo structure
2013-09-13 11:08:55 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
rtpbin: RTPArrivalStats -> RTPPacketInfo
Rename a structure because we are also going to use this for the sender
bits.
2013-09-13 10:55:31 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
source: small cleanups
2013-09-12 13:31:01 -0300 Thiago Santos <thiago sousa santos collabora com>
* gst/isomp4/qtdemux.c:
qtdemux: only update stop position if seek requests it
Check for GST_SEEK_TYPE_NONE for stop poistion and only update
the stop time if it is requested. Otherwise just maintain whatever
was stored at the segment
https://bugzilla.gnome.org/show_bug.cgi?id=707530
2013-09-13 08:53:25 +0200 Rico Tzschichholz <ricotz ubuntu com>
* gst/rtp/Makefile.am:
rtp: Add missing headers tp fix make dist
In addition to a956a6ceb2deb87cc1361aee1d6626449f46dab2
2013-09-12 15:07:48 +0200 Sebastian Dröge <slomo circular-chaos org>
* gst/audioparsers/gstflacparse.c:
flacparse: Make sure we have enough data to read image tags
Thanks to iputinei for reporting this on IRC.
2013-09-12 15:01:36 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: handle segments with non-0 start
We keep the DTS and PTS in running-time inside the jitterbuffer. Make sure to
transform it back to a buffer timestamp before pushing out the buffer.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707931
2013-09-11 13:11:58 -0600 Seán de Búrca <leftmostcat gmail com>
* gst/matroska/matroska-demux.c:
matroskademux: Fix off-by-one in validation of UTF-8
https://bugzilla.gnome.org/show_bug.cgi?id=707933
2013-09-11 14:32:17 -0300 Thibault Saunier <thibault saunier collabora com>
* gst/videomixer/videomixer2.c:
videomixer: Do not check if caps are empty when they are NULL
In the case the caps are actually NULL, we should just concider it the
same way as empty caps in that case.
2013-09-10 16:44:53 -0600 Seán de Búrca <leftmostcat gmail com>
* gst/videomixer/blendorc-dist.c:
* gst/videomixer/blendorc-dist.h:
* gst/videomixer/videomixerorc-dist.c:
* gst/videomixer/videomixerorc-dist.h:
videomixer: fix build if orc is not installed
https://bugzilla.gnome.org/show_bug.cgi?id=707886
2013-09-10 17:57:49 -0300 Thiago Santos <thiago sousa santos collabora com>
* gst/matroska/matroska-demux.c:
matroskademux: Preserve seqnum when pushing seek upstream
After converting a seek from time to bytes, use the same seqnum
on the event that goes upstream
2013-09-05 00:17:16 -0300 Thiago Santos <thiago sousa santos collabora com>
* gst/isomp4/qtdemux.c:
qtdemux: track streams that are EOS on push mode to finish earlier
When the segment has a defined stop position, qtdemux should check
when streams reach this position and mark those as EOS. When all
streams are EOS it will return GST_FLOW_EOS to upstream to allow
the pipeline to finish instead of continuously consume buffers
from upstream that are not useful for the segment.
https://bugzilla.gnome.org/show_bug.cgi?id=707530
2013-09-04 15:34:35 -0300 Thiago Santos <thiago sousa santos collabora com>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
qtdemux: preserve stop of segment when doing seeks in push mode
When handling seeks in push mode, qtdemux converts the seek to bytes
and pushes upstream. It needs to keep track of the seek and the
subsequent segment to be able to map them back to the requested
seek time and properly preserve the segment stop of the seek.
This is done by using the start offset in bytes of the seek,
that should be the same of the segment from upstream. And this
is also backwards compatible with what qtdemux already was using.
https://bugzilla.gnome.org/show_bug.cgi?id=707530
2013-07-26 19:40:53 +0200 Mathieu Duponchelle <mathieu duponchelle epitech eu>
* gst/videomixer/videomixer2.c:
* gst/videomixer/videomixer2pad.h:
videomixer: Add colorspace conversion
https://bugzilla.gnome.org/show_bug.cgi?id=704950
2013-08-06 15:38:39 +0200 Mathieu Duponchelle <mathieu duponchelle epitech eu>
* gst/videomixer/videomixer2.c:
videomixer: Don't send reconfigure event when formats or PAR are different
It is racy with multiple pads.
https://bugzilla.gnome.org/show_bug.cgi?id=704950
2013-07-25 13:49:57 +0200 Mathieu Duponchelle <mathieu duponchelle epitech eu>
* gst/videomixer/Makefile.am:
* gst/videomixer/blend.c:
* gst/videomixer/blendorc.orc:
* gst/videomixer/gstcms.c:
* gst/videomixer/gstcms.h:
* gst/videomixer/videoconvert.c:
* gst/videomixer/videoconvert.h:
* gst/videomixer/videomixer2.c:
* gst/videomixer/videomixerorc.orc:
videomixer: Bundle private copies of videoconvert code
Ideally, this would be part of libgstvideo.
Prefixes videoconvert symbols with videomixer_.
https://bugzilla.gnome.org/show_bug.cgi?id=704950
2013-08-22 00:03:48 +0200 Mathieu Duponchelle <mathieu duponchelle epitech eu>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2: Use newly #defined metadata names.
2013-09-09 15:11:51 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: only wait if we flushed
Only wait for the STREAM_LOCK when we flushed something when sending
a command for PAUSED or PLAYING.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707611
2013-09-09 15:09:41 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: return when a flush was issued
Make gst_rtspsrc_loop_send_cmd() return TRUE when the current
action has been flushed
2013-09-09 11:16:40 +0200 David Holroyd <dave badgers-in-foil co uk>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c:
* gst/rtp/gstrtpL24depay.c:
* gst/rtp/gstrtpL24depay.h:
* gst/rtp/gstrtpL24pay.c:
* gst/rtp/gstrtpL24pay.h:
* tests/check/elements/rtp-payloading.c:
rtp: add L24 pay and depayloader
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734
2013-09-09 14:46:42 +0200 Sebastian Dröge <slomo circular-chaos org>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: Fix missing condition in previous commit
2013-09-09 14:44:58 +0200 Sebastian Dröge <slomo circular-chaos org>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: Also fix strides for other semi-planar video formats
2013-09-09 14:41:42 +0200 Andreea Fulger <andreea fulger parrot com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: Fix stride for NV12/NV21
https://bugzilla.gnome.org/show_bug.cgi?id=707758
2013-09-07 16:37:03 +0200 Matej Knopp <matej knopp gmail com>
* gst/matroska/matroska-read-common.c:
matroskademux: fix leaking buffer and caps
https://bugzilla.gnome.org/show_bug.cgi?id=707688
2013-09-05 19:46:37 +0100 Tim-Philipp Müller <tim centricular net>
* gst/udp/gstudpsrc.c:
udpsrc: fix build on win32
gstudpsrc.c:855:15: error: #if with no expression
2013-09-04 15:50:42 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/avi/gstavidemux.c:
avidemux: handle unseekable streams
Handle streams that we can't seek in and ignore them in the
seek logic.
2013-09-04 15:25:39 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/avi/gstavidemux.c:
avidemux: only check video compression for video streams
Or else we might deref a stream with a NULL strf.vids and segfault
2013-06-18 13:27:20 +0100 Alex Ashley <bugzilla ashley-family net>
* gst/isomp4/atoms.c:
* gst/isomp4/fourcc.h:
* gst/isomp4/ftypcc.h:
* gst/isomp4/gstrtpxqtdepay.c:
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux_fourcc.h:
* gst/isomp4/qtdemux_types.c:
qtdemux: Add support for the avc3 sample entry format of the AVC file format
Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new
structure for fragmented MP4 called "avc3". The principal difference
between AVC1 and AVC3 is the location of the codec initialisation
data (e.g. SPS, PPS). In AVC1 this data is placed in the initial
MOOV box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data
goes in the first sample of every fragment (i.e. the first sample in
each mdat box). The principal reason for avc3 is to make it easier
for client implementations, because it removes the requirement to
insert the SPS+PPS in to the decoder pipeline every time there is a
representation change.
This commit adds support for the "avc3" atom, which is almost identical
to the "avc1" atom, except it does not contain any SPS or PPS data.
https://bugzilla.gnome.org/show_bug.cgi?id=702004
2013-09-04 00:27:50 +0200 Mathieu Duponchelle <mathieu duponchelle epitech eu>
* gst/videomixer/videomixer2.c:
videomixer: Don't set EOS to FALSE when the collectpad *is* EOS
https://bugzilla.gnome.org/show_bug.cgi?id=707238
2013-09-03 17:32:41 +0200 Matej Knopp <matej knopp gmail com>
* gst/audioparsers/gstflacparse.c:
flacparse: cleanup on error after state change
https://bugzilla.gnome.org/show_bug.cgi?id=707229
2013-09-03 11:23:24 +0200 Sebastian Dröge <slomo circular-chaos org>
* gst/udp/gstudpsrc.c:
* gst/udp/gstudpsrc.h:
udpsrc: Bind to multicast addresses on non-Windows systems
On Windows it's not possible to bind to a multicast address
but the OS will make sure to filter out all packets that
arrive not for the multicast address the socket joined.
On Linux and others it is necessary to bind to a multicast
address to let the OS filter out all packets that are received
on the same port but for different addresses than the multicast
address
And deprecate the multicast-group property and replace it with the
address property.
https://bugzilla.gnome.org/show_bug.cgi?id=707042
2013-09-03 10:10:01 +0200 Matej Knopp <matej knopp gmail com>
* gst/audioparsers/gstflacparse.c:
flacparse: Free GstBaseParseFrame if pushing a header failed
2013-09-02 16:02:37 +0200 Sebastian Dröge <slomo circular-chaos org>
* gst/udp/gstudpsrc.c:
udpsrc: Refactor address resolval into its own function
2013-09-02 23:00:29 +0100 Tim-Philipp Müller <tim centricular net>
* gst/replaygain/gstrganalysis.c:
replaygain: fix taglist leak in rganalysis
And add some FIXMEs.
2013-09-02 22:50:58 +0100 Tim-Philipp Müller <tim centricular net>
* tests/check/elements/rganalysis.c:
tests: rganalysis: rename function for clarity
2013-03-18 14:32:07 +0100 Christoph Reiter <reiter christoph gmail com>
* tests/check/elements/rganalysis.c:
tests: fix skipped rganalysis tests
In 0.10 elements would post tag messages on the bus
directly, and rganalysis would only post a tag message
when it changed tags. In 1.0, only sinks post tag
messages when they receive the serialised tag event.
This means that we get an additional tag message on
the bus now where we didn't expect one before.
https://bugzilla.gnome.org/show_bug.cgi?id=695090
2013-09-02 11:46:52 +0200 Sebastian Dröge <slomo circular-chaos org>
* gst/audioparsers/gstflacparse.c:
flacparse: Properly propagate downstream flow returns upstream
https://bugzilla.gnome.org/show_bug.cgi?id=707229
2013-09-01 21:18:38 +0100 Tim-Philipp Müller <tim centricular net>
* ext/shout2/gstshout2.c:
* gst/avi/gstavi.c:
* gst/isomp4/isomp4-plugin.c:
* gst/rtsp/gstrtsp.c:
* sys/sunaudio/gstsunaudio.c:
* sys/v4l2/gstv4l2.c:
Don't use setlocale in plugins()
Only apps should call setlocale(), not libraries.
2013-08-29 13:15:15 +0200 Wim Taymans <wim taymans collabora co uk>
* gst/rtp/gstrtpmpvpay.c:
rtpmpvpay: Fix RTP buffer allocation in rtpmpvpay
RTP buffer allocation should not be done with padding for the specific MPEG2
header as the padding is done at the end of the buffer and the last byte is
the size of the padding.
https://bugzilla.gnome.org/show_bug.cgi?id=706970
2013-08-28 10:51:32 +0200 Bernhard Miller <bernhard miller streamunlimited com>
* gst/autodetect/gstautovideosink.c:
* gst/autodetect/gstautovideosink.h:
autovideosink: add sync property
https://bugzilla.gnome.org/show_bug.cgi?id=706955
2013-08-28 07:15:00 +0200 Bernhard Miller <bernhard miller streamunlimited com>
* gst/autodetect/gstautoaudiosink.c:
* gst/autodetect/gstautoaudiosink.h:
autoaudiosink: introduce sync property
https://bugzilla.gnome.org/show_bug.cgi?id=706955
2013-08-27 17:33:40 -0300 Thiago Santos <thiago sousa santos collabora com>
* gst/isomp4/qtdemux.c:
qtdemux: push buffers after segment stop until reaching a keyframe
This should make decoders able to precisely push buffers until the stop
time in case they need the next keyframe to do it.
Also, according to gst_segment_clip, it should only push a buffer that
the starting ts is strictly smaller than the segment stop, so we change
the min < comparison for <=
2013-08-28 13:26:47 +0200 Sebastian Dröge <slomo circular-chaos org>
* configure.ac:
Back to development
Download
========
http://download.gnome.org/sources/gst-plugins-good/1.1/gst-plugins-good-1.1.90.tar.xz (2.73M)
sha256sum: b113cf39d96c5977fbe1eef9c3ce05f5b4be797038fdb1f91971e0cbe6f78197
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