ChangeLog
=========
2013-07-29 Sebastian Dröge <sebastian.dro...@collabora.co.uk>
* configure.ac:
releasing 1.1.3
2013-07-29 12:12:41 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/avi/gstavidemux.c:
* gst/flv/gstflvdemux.c:
* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
gst: Don't swap start/stop for negative rates in the SEGMENT query
2013-07-29 11:18:40 +0200 Matej Knopp <matej.kn...@gmail.com>
* gst/isomp4/qtdemux.c:
qtdemux: Check for data size when parsing h264 codec data from strf
atom
2013-07-29 10:53:54 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/matroska/matroska-demux.c:
matroskademux: Implement SEGMENT query
2013-07-29 10:53:47 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/flv/gstflvdemux.c:
flvdemux: Implement SEGMENT query
2013-07-29 10:50:59 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/avi/gstavidemux.c:
avidemux: Implement SEGMENT query
2013-07-27 18:10:22 +0200 Matej Knopp <matej.kn...@gmail.com>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux_fourcc.h:
qtdemux: Support H264 fourcc
https://bugzilla.gnome.org/show_bug.cgi?id=704996
2013-07-28 18:09:33 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* ext/flac/gstflacenc.c:
flacenc: Fix handling of image tags
The caps should be used to get the mimetype and there is
only an info structure for the GstSample if the image-type
is not NONE.
2013-07-28 18:04:32 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* ext/flac/gstflacenc.c:
flacenc: Don't crash if there is no image tag information
https://bugzilla.gnome.org/show_bug.cgi?id=705018
2013-07-28 17:38:56 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/avi/gstavidemux.c:
avidemux: Fix duration reporting in push mode
https://bugzilla.gnome.org/show_bug.cgi?id=700933
2013-07-28 17:32:27 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/avi/gstavidemux.c:
avidemux: Don't forget unmapping and unreffing buffer
2013-07-26 21:06:17 +0200 Matej Knopp <matej.kn...@gmail.com>
* gst/avi/gstavidemux.c:
avidemux: unmap buffer
https://bugzilla.gnome.org/show_bug.cgi?id=704951
2013-07-26 22:31:41 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: don't make buffer writable prematurely
There is no reason to make the SR buffer writable at this point. This
is better
delayed until needed.
2013-07-26 22:25:50 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: ignore RTCP for inactive sources
2013-07-26 22:25:17 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: small cleanup
2013-07-26 17:17:31 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
* gst/rtpmanager/rtpsource.h:
session: handle partial RTCP report blocks
When we have more SSRCs to report than what fit in an RTCP packet,
use a
generation counter to make sure all of them end up in a packet
eventually.
2013-07-26 17:23:10 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: create SSRC before doing session cleanup
Make the internal source before we do session cleanup
2013-07-26 17:21:08 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: reorganize the report block code
2013-07-26 16:02:01 +0200 Matej Knopp <matej.kn...@gmail.com>
* gst/matroska/matroska-demux.c:
matroskademux: fix memory leak in check_subtitle_buffer
https://bugzilla.gnome.org/show_bug.cgi?id=704921
2013-07-26 14:21:40 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: refactor active and sender checks
2013-07-26 12:06:35 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: remove internal sources on timeout
When an internal source times out and becomes a receiver, remove it.
2013-07-26 11:47:56 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: create an internal source for RTCP
When we need to do RTCP and we don't have an internal source yet,
make one.
2013-07-26 10:47:28 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
* gst/rtpmanager/rtpsource.c:
session: remove old code to change SSRC
Remove code used to change the SSRC after a collision. We now send
a RECONFIGURE event upstream to make the upstream element change the
SSRC.
2013-07-26 10:42:44 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsource.c:
source: don't update packet SSRC
Remove the code to update the SSRC in packets, it can never be called
now that
we always use a source with matching packet SSRC.
2013-07-26 10:24:22 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
session: delay allocation of internal source
Allocate the internal source when we receive a caps with the SSRC or
when we see
a buffer with the SSRC.
2013-07-26 10:00:58 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/rtpsession.c:
session: generate reconfigure on collision
When we detect a collision, change the SSRC that we suggest upstream
and trigger RECONFIGURE. This should make upstream select a new SSRC.
2013-07-26 09:37:24 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
session: produce RTCP for all internal sources
Loop over all the internal sources and produce RTCP. We also need
to queue the RTCP packets and send them when we are finished.
2013-07-26 01:40:20 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
session: deprecate internal source and ssrc properties
Deprecate the internal source and internal ssrc properties. There
might
be more than one internal source.
2013-07-26 01:29:08 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: internal sources don't use probation
2013-07-26 01:24:07 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/rtpsession.c:
session: give caps to session
Let the session parse the caps and update its SSRC when needed.
2013-07-26 01:14:04 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
session: make method to suggest available SSRC
Make a method to suggest the best available SSRC. This is the SSRC of
the last
created internal source and is used to instruct upstream to produce
this
SSRC.
2013-07-26 01:01:49 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
session: keep SDES and set on new internal sources
Keep track of the SDES ourselves and set it on all newly created
internal sources.
2013-07-26 00:48:25 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: make method to make internal sources
Add a method to obtain an internal source and use it to create
our internal source
2013-07-26 00:29:41 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpstats.h:
session: count internal sources and how many are senders
2013-07-26 00:14:29 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
rtpsession: separate BYE marking and scheduling
First mark sources with BYE and then schedule the BYE RTCP message.
2013-07-25 23:56:46 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: get SSRC from RTCP packet itself
Get the SSRC from the RTCP packet instead.
2013-07-25 23:51:34 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: fix bandwidth calculation
We iterate over all sources and the internal one is also in the
hashtable so avoid adding it twice.
2013-07-25 23:38:08 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: add some docs
2013-07-25 23:11:05 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: Rearrange RTCP reporting a little
Make a function to generate an RTCP packet for a source, pass the
source as a
parameter.
Move timeout of collisions to session cleanup phase.
2013-07-25 22:39:04 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: move check for is_early around
Move the check for the early RTCP to where it is needed and used.
2013-07-25 17:35:02 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: parse packet outside of the session lock
2013-07-25 17:34:06 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: do nicer checks for internal sources
2013-07-25 17:15:37 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
session: let source keep track if it sent BYE
2013-07-25 17:06:22 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsource.c:
source: reset more
2013-07-25 16:49:41 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
source: also use the source for bye_reason
Store the BYE reason in our internal source object. Rename the
methods on the
source object a little because now the BYE can be received in RTCP or
set when the session wants to send BYE.
2013-07-25 16:24:04 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
session: configure sdes with structure only
Remove code to configure the SDES with methods and types, only
allow configuration with GstStructure
2013-07-25 15:56:39 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: refactor add and find source
Make functions to find and add a source to the hashtable.
2013-07-25 15:43:11 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
session: remove source from sync_rtcp
We don't need to know the sender source of the session in the
callback, the SR packet is for all participants in the session.
2013-07-24 14:18:14 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: add some more debug
2013-07-15 17:11:45 +0100 Vincent Penquerc'h
<vincent.penque...@collabora.co.uk>
* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstaacparse.h:
aacparse: allow conversion from ADTS to raw AAC
Some muxers (eg, qtmux) only support raw AAC, so this allows linking
an encoder that outputs ADTS only to those muxers.
The conversion is simple (omit the first 7 or 9 bytes of the frame),
but has to be done in pre_push instead of handle_frame as 1.0 does
not seem to allow skipping bytes there as 0.10 used to.
Other conversions are not supported (yet).
2013-07-15 17:15:44 +0100 Vincent Penquerc'h
<vincent.penque...@collabora.co.uk>
* gst/audioparsers/gstaacparse.c:
aacparse: fix object_type parsing off-by-one in ADTS frame
According to http://wiki.multimedia.cx/index.php?title=ADTS,
the value stored in ADTS headers is one less than the object
type of the AAC stream.
A look at ffmpeg shows it also adds 1 to the value read off
the ADTS header.
Note that this might break other things that happen to have
an inverse off by one to match the existing code.
2013-07-25 11:13:01 -0300 Thiago Santos <thiago.sousa.san...@collabora.com>
* gst/avi/gstavidemux.c:
avidemux: fix seqnum handling for seeks
Use the same seqnum as the seek for flushes/segments that are
caused by the seek. Also do the same for segment events
Fixes #676242
2013-07-25 01:39:58 -0300 Thiago Santos <thiago.sousa.san...@collabora.com>
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-demux.h:
matroskademux: fix seqnum handling for seeks
Use the same seqnum as the seek for flushes/segments that are
caused by the seek. Also do the same for segment events
Fixes #676242
2013-07-25 01:11:31 -0300 Thiago Santos <thiago.sousa.san...@collabora.com>
* gst/isomp4/qtdemux.c:
qtdemux: correctly handle seqnum for seeks and segments
Use the same seqnum on messages and events for derived events.
Fixed for flushes / stream-start / segment after a seek, and segment
after a segment.
Fixes #676242
2013-07-12 20:01:42 +0200 Arnaud Vrac <av...@freebox.fr>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: always ignore HEAD errors
https://bugzilla.gnome.org/show_bug.cgi?id=704241
2013-07-25 14:26:07 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* ext/jpeg/gstjpegenc.c:
jpegenc: Clean up reset/start/stop handling
2013-07-25 14:13:10 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegdec.h:
jpegdec: Use base class error handling function instead of
replicating it here
2013-07-25 14:12:56 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* ext/jpeg/gstjpegdec.c:
jpegdec: Clean up handling of reset/start/stop
2013-07-25 10:41:22 +0100 Tim-Philipp Müller <tim.mul...@collabora.co.uk>
* tests/files/id3-407349-1.tag:
* tests/files/id3-407349-2.tag:
* tests/files/id3-447000-wcop.tag:
tests: fix test ID3 tags up not to rely on dodgy typefinding code
Change 0xff 0xfb 'mp3' marker to 'fLaC' marker, so we can fix
the typefinder.
https://bugzilla.gnome.org/show_bug.cgi?id=681368
2013-07-25 08:22:45 +0200 Alessandro Decina <alessandr...@gmail.com>
* sys/osxaudio/gstosxaudiosink.c:
osxaudiosink: intersect the probed caps with the filter passed to
get_caps()
2013-07-24 14:17:45 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/gstrtpbin.c:
bin: fix compilation
2013-07-24 12:42:31 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtp/gstrtpvrawdepay.c:
vrawdepay: fix UYVP format
2013-07-24 12:41:58 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtp/gstrtpvrawpay.c:
vrawpay: fix UYVP format
2013-07-24 12:41:44 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtp/gstrtpvrawpay.c:
vrawpay: fix caps
2013-07-24 10:49:03 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: fix locking
Take the lock earlier so that we do things that follow with the right
locking.
2013-07-23 17:40:02 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
rtpsession: don't use invalid times in RTCP timeouts
An invalid timeout can be calculated when we disabled RTCP by setting
the
bandwidth to 0. Make sure all code can handle this case.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674626
2013-07-23 17:38:20 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
rtpsession: lock session when changing bandwidth
Take the session lock when changing the bandwidth properties so that
we don't
end up with inconsistent behaviour.
2013-07-23 17:37:05 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/rtpmanager/rtpsession.c:
session: reset some RTCP variables
The early_send time was set to 0 and always triggering an early RTCP
packet.
2013-07-23 15:03:31 +0200 Edward Hervey <edw...@collabora.com>
* gst/isomp4/qtdemux.c:
qtdemux: Add all the mpeg XDCAM variants
This should cover all known XDCAM variants (which are all mpeg2 video)
Fixes #672227
2013-07-03 18:41:42 +0200 Carlos Rafael Giani <d...@pseudoterminal.org>
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpbin.h:
rtpbin: added custom downstream sync event
rtpbin can now send a custom in-band downstream event which informs
downstream that the bin has received an RTCP SR packet. This is useful
for applications which want to drop the initial unsynchronized
received
RTP packets.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560
Signed-off-by: Carlos Rafael Giani <d...@pseudoterminal.org>
2013-07-22 18:00:16 +0100 Tim-Philipp Müller <tim.mul...@collabora.co.uk>
* gst/deinterlace/gstdeinterlace.c:
deinterlace: fix on-the-fly changing of "mode" and "fields" properties
We call setcaps() to reconfigure ourselves, but we need to pass
the current *sink* caps, not the source caps then. Also fix a
caps leak.
https://bugzilla.gnome.org/show_bug.cgi?id=641599
2013-07-22 15:23:39 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/wavparse/gstwavparse.c:
wavparse: Add support for group-id in the stream-start event
2013-07-22 15:23:20 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Add support for group-id in the stream-start event
2013-07-22 15:23:11 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/rtpmanager/gstrtpsession.c:
rtpsession: Add support for group-id in the stream-start event
2013-07-22 15:22:55 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-demux.h:
matroskademux: Add support for group-id in the stream-start event
2013-07-22 15:22:47 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
qtdemux: Add support for group-id in the stream-start event
2013-07-22 15:22:36 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/flv/gstflvdemux.c:
* gst/flv/gstflvdemux.h:
flvdemux: Add support for group-id in the stream-start event
2013-07-22 15:22:16 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/avi/gstavidemux.c:
* gst/avi/gstavidemux.h:
avidemux: Add support for group-id in the stream-start event
2013-07-22 15:21:49 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* ext/dv/gstdvdemux.c:
* ext/dv/gstdvdemux.h:
dvdemux: Add support for group-id in the stream-start event
2013-07-19 22:59:15 +0200 Mathieu Duponchelle <mathieu.duponche...@epitech.eu>
* gst/videomixer/videomixer2.c:
videomixer: use gst_util_uint64_scale*_round.
There could be a case where:
1) you do a new set_caps after buffers have been processed.
2) ts_offset gets set to a different value, eg 0.033333333
3) your pads get EOS, but the check dor that doesn't work
because you use ts_offset + a truncated value < segment.stop
4) so in the next collected, you end up comparing for example:
0.9999999999 > 1., which is false and means you don't send EOS.
Also adds scale_round in two other places where it potentially could
have caused problems.
2013-07-15 17:55:19 -0400 Olivier Crête <olivier.cr...@collabora.com>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux_fourcc.h:
qtdemux: Add WRLE support
2013-07-19 19:35:26 +0100 Tim-Philipp Müller <tim.mul...@collabora.co.uk>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux_fourcc.h:
qtdemux: make files from Vivotek camera play
Skip tracks of 'vivo' subtype with empty stsd instead of
erroring out saying that the file is broken.
https://bugzilla.gnome.org/show_bug.cgi?id=699791
2013-07-19 17:14:06 +0100 Tim-Philipp Müller <tim.mul...@collabora.co.uk>
* gst/isomp4/gstqtmux.c:
qtmux: when streaming don't try to seek when stopping
It might cause errors in sinks that are not seekable and
have reported this (like e.g. fdsink)
https://bugzilla.gnome.org/show_bug.cgi?id=696228
2013-07-19 17:26:54 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/isomp4/qtdemux.c:
qtdemux: simplify some helpers
Some helper functions are not needed anymore or can be simplified.
2013-07-19 17:12:37 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/isomp4/qtdemux.c:
qtdemux: for non-raw video, move palette in caps
We only need to append the palette to raw video buffers, non-raw
video has the
palette in the caps still.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292
2013-07-19 01:49:20 +0200 Arnaud Vrac <av...@freebox.fr>
* gst/isomp4/qtdemux.c:
qtdemux: nitpicking in esds parsing
2013-07-19 01:49:07 +0200 Arnaud Vrac <av...@freebox.fr>
* gst/isomp4/qtdemux.c:
qtdemux: set proper caps for mpeg-1 audio
Remove AAC specific fields from mpeg-1 audio caps, remove assumption
that the mpeg1 audio layer is 3, and set `parsed' field.
https://bugzilla.gnome.org/show_bug.cgi?id=704548
2013-06-17 21:27:37 +0200 Arnaud Vrac <av...@freebox.fr>
* ext/vpx/gstvp8dec.h:
* ext/vpx/gstvp8enc.h:
* ext/vpx/gstvp9dec.h:
* ext/vpx/gstvp9enc.h:
vpx: fix compilation when encoder or decoder headers are not installed
https://bugzilla.gnome.org/show_bug.cgi?id=704547
2013-07-16 20:41:15 -0400 Nicolas Dufresne <nicolas.dufre...@collabora.com>
* tests/check/elements/videocrop.c:
videocrop: Fix unit for GRAY16 formats
2013-07-16 22:17:17 +0200 Arnaud Vrac <av...@freebox.fr>
* gst/isomp4/qtdemux.c:
qtdemux: remove chapter stream
Remove all streams that are actually table of contents, since we will
never need the data after parsing them.
2013-07-16 21:59:37 +0200 Arnaud Vrac <av...@freebox.fr>
* gst/isomp4/qtdemux.c:
qtdemux: send gap event for sparse streams in push mode
This allows to pre-roll at least if the next subtitle buffer
is far away.
2013-07-16 21:56:07 +0200 Arnaud Vrac <av...@freebox.fr>
* gst/isomp4/qtdemux.c:
qtdemux: do not use indexes from sparse stream when seeking in push
mode
This makes seeking more accurate in push mode, since the previous
keyframe on a sparse stream might be far away.
2013-07-16 21:04:07 +0200 Arnaud Vrac <av...@freebox.fr>
* gst/isomp4/qtdemux.c:
qtdemux: advertise subtitle streams as sparse
2013-07-17 17:11:44 +0200 Arnaud Vrac <av...@freebox.fr>
* gst/matroska/matroska-demux.c:
mastrokademux: do not push discont buffers if they aren't discont
Unset the discont flag instead of posssibly pushing a buffer with
a flag that's still set.
https://bugzilla.gnome.org/show_bug.cgi?id=682110
2013-07-17 15:10:00 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/isomp4/qtdemux.c:
qtdemux: extract the palette from stsd
Sometimes a palette is inside the stsd, extract it instead of always
using
the default one
2013-07-17 14:30:16 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/goom2k1/gstgoom.c:
goom2k1: Fix event handling and negotiate as soon as possible
2013-07-17 14:27:57 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* gst/goom/gstgoom.c:
goom: Fix event handling and negotiate as soon as possible
2013-07-11 19:45:17 +0200 Andoni Morales Alastruey <ylat...@gmail.com>
* sys/osxvideo/osxvideosink.m:
osxvideosink: warn about the future deprecation of the "embed"
property
2013-07-17 09:56:01 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/isomp4/qtdemux.c:
qtdemux: add support for WRAW
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292
2013-07-17 09:54:58 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* gst/isomp4/qtdemux.c:
qtdemux: palette is appended to buffers, not in caps
Fix the palette handling, in 1.0 we append the palette to the buffer
instead of
placing it on the caps.
See also https://bugzilla.gnome.org/show_bug.cgi?id=704292
2013-07-16 15:37:49 -0400 Olivier Crête <olivier.cr...@collabora.com>
* gst/rtp/gstrtpgstpay.c:
* gst/rtp/gstrtpmp2tpay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtpmpvpay.c:
rtp: Use gst_adapter_take_buffer_fast() where possible in RTP
payloaders
2013-07-15 16:24:07 +0200 Arnaud Vrac <av...@freebox.fr>
* gst/isomp4/qtdemux.c:
qtdemux: reset segment on flush stop
cca2f555d14 introduces a regression, where the demux segment is not
reset on flush stop, so the next upstream segment event will calculate
an invalid base time on the new segment to be sent downstream.
https://bugzilla.gnome.org/show_bug.cgi?id=704255
2013-07-06 17:20:49 +0200 Matej Knopp <matej.kn...@gmail.com>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
qtdemux: offset samples according to edit list
https://bugzilla.gnome.org/show_bug.cgi?id=700264
2013-07-14 12:50:13 +1200 Douglas Bagnall <doug...@halo.gen.nz>
* tests/examples/spectrum/spectrum-example.c:
level: Fix the spectrum example for 1.0
The "message" property has been replaced by "post-messages".
Pre-patch output:
(test_spectrum:23101): GLib-GObject-WARNING **: g_object_set_valist:
object class `GstSpectrum' has no property named `message'
New spectrum message, endtime 0:00:00.100000000
(test_spectrum:23101): GStreamer-CRITICAL **:
gst_value_list_get_value: assertion `GST_VALUE_HOLDS_LIST (value)'
failed
[...]
Post-patch:
New spectrum message, endtime 0:00:00.100000000
band 0 (freq 400): magnitude -65.988777 dB phase 1.533397
band 1 (freq 1200): magnitude -65.545563 dB phase -0.780900
band 2 (freq 2000): magnitude -64.791946 dB phase -0.799611
band 3 (freq 2800): magnitude -64.556175 dB phase -0.063615
[...]
https://bugzilla.gnome.org/show_bug.cgi?id=704179
2013-07-13 20:56:26 +0200 Matej Knopp <matej.kn...@gmail.com>
* gst/audioparsers/gstaacparse.c:
aacparse: be less verbose when parsing LOAS streams
https://bugzilla.gnome.org/show_bug.cgi?id=704162
2013-07-12 12:31:39 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* ext/pulse/pulsesink.h:
sink: alaw/mulaw caps don't have a layout property
2013-07-12 12:27:53 +0200 Wim Taymans <wim.taym...@collabora.co.uk>
* ext/pulse/pulseutil.c:
pulse: relax mulaw and alaw format checks
The audio library considers them as encoded formats and does not fill
in the
sample width. The audio ringbuffers identifies the format as
alaw/mulaw and that
is always 8 bits.
2013-07-11 16:13:05 +0200 Matej Knopp <matej.kn...@gmail.com>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
* gst/isomp4/qtdemux_fourcc.h:
* gst/isomp4/qtdemux_types.c:
qtdemux: unselect instead of ignoring disabled track, detect chapter
track
https://bugzilla.gnome.org/show_bug.cgi?id=704007
2013-07-11 20:41:23 -0300 Thiago Santos <thiago.sousa.san...@collabora.com>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: ignore errors from HEAD request
HEAD requests are used to check the server headers to see if it
seekable. Ignore errors from those requests as they shouldn't be
critical.
https://bugzilla.gnome.org/show_bug.cgi?id=704053
2013-07-12 03:24:08 +0800 Kyosuke Nekomura <supercatexp...@gmail.com>
* gst/audiofx/audioecho.c:
audioecho: Fix handling of delay property in PLAYING/PAUSED state
https://bugzilla.gnome.org/show_bug.cgi?id=703901
2013-07-09 17:56:57 -0400 Olivier Crête <olivier.cr...@collabora.com>
* gst/rtpmanager/gstrtpmux.c:
rtpmux: Enable proxy caps on the src pads
2013-07-11 16:57:15 +0200 Sebastian Dröge <sl...@circular-chaos.org>
* configure.ac:
Back to development
Download
========
http://download.gnome.org/sources/gst-plugins-good/1.1/gst-plugins-good-1.1.3.tar.xz
(2.70M)
sha256sum: 1f516cb39e24ba5544a503bef246f9cf5b002dfbcbc2c475c6b8d5c23bc140a6
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