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=== release 1.6.0 ===
2015-09-25 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.6.0
2015-09-25 14:08:09 +0200 Thibault Saunier <tsaunier@gnome.org>
* gst/smpte/gstsmptealpha.c:
smptealpha: Do not set width/height before comparing with old values
Otherwise we end up considering the values did not change and we wrongly
work with the old video format (which will lead to wrong
behaviour/segfaults).
https://bugzilla.gnome.org/show_bug.cgi?id=755621
2015-09-23 20:59:00 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/isomp4/qtdemux.c:
qtdemux: Accumulate segments for edit lists before activating the next segment
eceb2ccc739092d964d78945e19c2ecedbd214e2 broke segment seeks by always
accumulating segments manually when activating a segment. This is only
needed when handling edit lists, not when activating a segment because of a
seek. Do the accumulation when switching edit list segments instead.
This fixes segment seeks again, while keeping edit lists playback working.
https://bugzilla.gnome.org/show_bug.cgi?id=755471
2015-09-23 17:43:51 +0530 Vikram Fugro <vikram.fugro@gmail.com>
* gst/spectrum/gstspectrum.c:
spectrum: send phase values in the GstMessage for Phase info
https://bugzilla.gnome.org/show_bug.cgi?id=755463
2015-09-22 00:46:01 +1000 Jan Schmidt <jan@centricular.com>
* gst/matroska/matroska-mux.c:
matroska-mux: Don't output a warning on MONO multiview mode.
2015-09-19 17:02:18 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtp/gstrtptheoradepay.c:
rtptheoradepay: Fix memory leaks
The same memory leaks were fixed in identical fashion for
vorbisdepay in 06efeff5d979576a252e5dae57f46d6445b1df12 in 2009.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
2015-09-19 17:04:07 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbispay.c:
rtp{vorbis,theora}{pay,depay}: Cosmetic cleanup
* use g_list_free_full(), don't iterate elements maually when freeing
* call gst_rtp_*_pay_clear_packet(), don't duplicate its code
* use gst_buffer_unref() to clarify that it is buffers being released,
instead of refering directly to gst_mini_object_unref()
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
2015-09-19 18:44:22 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbispay.c:
rtp{vorbis,theora}pay: Store headers in the packet buffers lists, not a NULL buffer
https://bugzilla.gnome.org/show_bug.cgi?id=755265
2015-09-18 19:33:13 +0200 Sebastian Dröge <sebastian@centricular.com>
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* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.signals:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-aasink.xml:
* docs/plugins/inspect/plugin-alaw.xml:
* docs/plugins/inspect/plugin-alpha.xml:
* docs/plugins/inspect/plugin-alphacolor.xml:
* docs/plugins/inspect/plugin-apetag.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-audioparsers.xml:
* docs/plugins/inspect/plugin-auparse.xml:
* docs/plugins/inspect/plugin-autodetect.xml:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-cacasink.xml:
* docs/plugins/inspect/plugin-cairo.xml:
* docs/plugins/inspect/plugin-cutter.xml:
* docs/plugins/inspect/plugin-debug.xml:
* docs/plugins/inspect/plugin-deinterlace.xml:
* docs/plugins/inspect/plugin-dtmf.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-effectv.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-flv.xml:
* docs/plugins/inspect/plugin-flxdec.xml:
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
* docs/plugins/inspect/plugin-goom.xml:
* docs/plugins/inspect/plugin-goom2k1.xml:
* docs/plugins/inspect/plugin-icydemux.xml:
* docs/plugins/inspect/plugin-id3demux.xml:
* docs/plugins/inspect/plugin-imagefreeze.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-isomp4.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mulaw.xml:
* docs/plugins/inspect/plugin-multifile.xml:
* docs/plugins/inspect/plugin-multipart.xml:
* docs/plugins/inspect/plugin-navigationtest.xml:
* docs/plugins/inspect/plugin-oss4.xml:
* docs/plugins/inspect/plugin-ossaudio.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-pulseaudio.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-rtpmanager.xml:
* docs/plugins/inspect/plugin-rtsp.xml:
* docs/plugins/inspect/plugin-shapewipe.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-smpte.xml:
* docs/plugins/inspect/plugin-soup.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speex.xml:
* docs/plugins/inspect/plugin-taglib.xml:
* docs/plugins/inspect/plugin-udp.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* docs/plugins/inspect/plugin-videobox.xml:
* docs/plugins/inspect/plugin-videocrop.xml:
* docs/plugins/inspect/plugin-videofilter.xml:
* docs/plugins/inspect/plugin-videomixer.xml:
* docs/plugins/inspect/plugin-vpx.xml:
* docs/plugins/inspect/plugin-wavenc.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
* docs/plugins/inspect/plugin-wavparse.xml:
* docs/plugins/inspect/plugin-ximagesrc.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* gst-plugins-good.doap:
* win32/common/config.h:
Release 1.5.91
2015-09-18 19:23:57 +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:
Update .po files
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2015-09-18 11:50:31 +0200 Sebastian Dröge <sebastian@centricular.com>
* po/zh_CN.po:
po: Update translations
2015-09-17 10:50:01 +0900 Eunhae Choi <eunhae1.choi@samsung.com>
* gst/avi/gstavidemux.c:
avidemux: Fix taglist leak
gst_tag_list_insert() does not take ownership of the inserted taglist.
https://bugzilla.gnome.org/show_bug.cgi?id=755138
2015-09-16 07:05:36 +1000 Jan Schmidt <jan@centricular.com>
* gst/audioparsers/gstaacparse.c:
aacparse: Skip LOAS AAC until a valid config is seen.
It's normal when dropping into the middle of a stream to
not always have the config available immediately, so skip LOAS
until a valid config is seen without either setting invalid
caps or erroring out.
https://bugzilla.gnome.org/show_bug.cgi?id=751386
2015-09-13 15:41:38 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: reset just a bit more upon flush_stop
2015-09-13 15:40:09 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: remove dead struct member
2015-09-11 17:09:28 +0900 Vineeth TM <vineeth.tm@samsung.com>
* gst/udp/gstmultiudpsink.c:
multiudpsink: fix GError memory leak when hostname resolution fails
https://bugzilla.gnome.org/show_bug.cgi?id=754869
2015-09-10 15:26:54 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/matroska/ebml-write.c:
matroskamux: drop HEADER flag from output buffers
Drop HEADER flag from output buffers if they are not indeed
headers.
Fixes resending of headers in tcp connection handling
https://bugzilla.gnome.org/show_bug.cgi?id=754768
2015-09-10 16:00:50 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/matroska/ebml-write.c:
matroskamux: fix matroskamux ! matroskademux
Don't carry over DISCONT flags from the input buffers to the
output buffer, or the demuxer might reset its state when it
receives the first data buffer just after parsing the simple
block header, and then expect sane data to follow.
Fixes matroskamux ! demux erroring out.
https://bugzilla.gnome.org/show_bug.cgi?id=754768
https://bugzilla.gnome.org/show_bug.cgi?id=657805
2015-09-09 12:51:40 -0700 Martin Kelly <martin@surround.io>
* gst/rtsp/README:
rtsp: fix small README typo
https://bugzilla.gnome.org/show_bug.cgi?id=754807
2015-09-04 19:45:37 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/audioparsers/gstwavpackparse.c:
wavpackparse: set both pts and dts so baseparse doesn't make up wrong dts after seeks
https://bugzilla.gnome.org/show_bug.cgi?id=752106
2015-09-04 19:34:41 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/audioparsers/gstflacparse.c:
flacparse: set both pts and dts so baseparse doesn't make up wrong dts after a seek
flac contains the sample offset in the frame header, so after a seek
without index flacparse will know the exact position we landed on and
timestamp buffers accordingly. It only set the pts though, which means
the baseparse-set dts which was set to the seek position prevails, and
since the seek was based on an estimate, there's likely a discrepancy
between where we wanted to land and where we did land, so from here on
that dts/pts difference will be maintained, with dts possibly multiple
seconds ahead of pts, which is just wrong. The easiest way to fix this
is to just set both pts and dts based on the sample offset, but perhaps
parsed audio should just not have dts set at all.
https://bugzilla.gnome.org/show_bug.cgi?id=752106
2015-09-06 16:33:02 +0100 Tim-Philipp Müller <tim@centricular.com>
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.signals:
docs: remove properties and signals that no longer exist
https://bugzilla.gnome.org/show_bug.cgi?id=726443
2013-10-11 15:13:00 +0000 George Chriss <gschriss@gmail.com>
* gst/flv/gstflvmux.c:
flvmux: Make the element count in arrays not include end
One-line removal of tags_written++
This should fix rtmp output to crtmpserver, and hopefully
noone is expecting that the element count includes the end
element, as different bits of documentation say different
things about whether it should or not.
https://bugzilla.gnome.org/show_bug.cgi?id=661624
2015-07-30 00:59:15 +1000 Jan Schmidt <jan@centricular.com>
* gst/flv/gstflvmux.c:
* gst/flv/gstflvmux.h:
flvmux: Store incoming bitrate tags and send in the metadata
Apparently the Microsoft Azure RTMP server requires that the
videodatarate and audiodatarate metadata be provided, so
set those, even if it's to 0. Use the actual input bitrate
tags if available.
2015-09-04 00:06:29 +1000 Jan Schmidt <jan@centricular.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Don't parse key data more than needed.
When an auxilliary streams are present in the SDP media,
there's no need to re-parse the SDP attributes multiple
times.
2015-09-03 20:56:55 +1000 Jan Schmidt <jan@centricular.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access.
In parse_keymgmt(), don't mutate the input string that's been passed
as const, especially since we might need the original value again if
the same key info applies to multiple streams (RTX, for example).
When a resource is 404, and we have auth info - retry with the auth
info the same as if we had receive unauthorised, in case the resource
isn't even visible until credentials are supplied.
Fix a memory leak handling Mikey data.
When generating a random keystring, don't overrun the 30 byte
buffer by generating 32 bytes into it.
2015-09-04 15:18:05 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/udp/gstudpsrc.c:
udpsrc: Fix build with GLib < 2.44
G_IO_ERROR_CONNECTION_CLOSED was added in 2.44.
2015-09-04 12:01:52 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/udp/gstudpsrc.c:
udpsrc: Ignore G_IO_ERROR_CONNECTION_CLOSED when receiving data
This happens on Windows if we use the same socket for sending packets,
and the remote sends ICMP port/host unreachable messages.
https://bugzilla.gnome.org/show_bug.cgi?id=754534
2015-09-02 21:12:41 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtp/gstrtptheoradepay.c:
* gst/rtp/gstrtpvorbisdepay.c:
rtpvorbis/theoradepay: Fix handling of fragmented packets
This was broken in b1089fb520 by not considering the full packet length of a
fragmented packet but only the length of the first one.
https://bugzilla.gnome.org/show_bug.cgi?id=754417
2015-09-01 15:39:22 -0400 Olivier Crête <olivier.crete@collabora.com>
* gst/dtmf/gstdtmfsrc.c:
* gst/dtmf/gstrtpdtmfsrc.c:
dtmfsrc: Reply to latency query
2015-08-31 16:42:30 -0400 Olivier Crête <olivier.crete@collabora.com>
* tests/check/elements/rtpsession.c:
tests: Fix rtpsession test failure
The time of the first RTCP packet is semi-random, so
sometimes it was produced before enough packets from
the second SSRC were received. First drop queued RTCP
packets, then advance the clock enough to ensure
that at least one new RTCP packet is produced.
https://bugzilla.gnome.org/show_bug.cgi?id=750731
2015-08-31 13:56:04 +0200 Stefan Sauer <ensonic@users.sf.net>
* tests/check/elements/level.c:
level: improve the test for multi-channel mode
Change the test to verify the read-index for multiple messages per buffer.
See https://bugzilla.gnome.org/show_bug.cgi?id=754144
2015-08-31 12:46:52 +0200 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
* gst/matroska/matroska-demux.c:
matroskademux: Align raw video frames to 32 bytes
Outputting unaligned video frames causes videoscale et al to
crash when attempting SIMD-accelerated conversion.
https://bugzilla.gnome.org/show_bug.cgi?id=736965
2015-08-26 23:16:46 +0200 Stefan Sauer <ensonic@users.sf.net>
* gst/level/gstlevel.c:
level: fix level calculations for mutliple channels
This was broken with 7b90bf32150897a141a29a12ecab555d8c5b7fab.
2015-08-27 10:28:55 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
* gst/smpte/gstsmpte.c:
smpte: Fix memory leak
In gst_smpte_collected(), check upfront if input formats are same
or not. This avoids allocation of in1 and in2 buffers and
subsequent memory leak when input formats do not match.
https://bugzilla.gnome.org/show_bug.cgi?id=754153
2015-08-21 11:52:19 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/elements/souphttpsrc.c:
tests: souphttpsrc: don't try to connect to dead radio server
2015-08-21 16:29:16 +0900 Vineeth TM <vineeth.tm@samsung.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Trivial fix to check correct condition
When checking for describe method, because of missing parentheses, wrong
condition is being checked, which will result in wrong behavior.
https://bugzilla.gnome.org/show_bug.cgi?id=753912
2015-08-21 13:19:02 +0900 Vineeth TM <vineeth.tm@samsung.com>
* gst/matroska/matroska-read-common.c:
matroska: read: fix tag list memory leak
gst_toc_entry_merge_tags makes a new ref of the taglist, so it should
be unref'ed as soon as the tags are merged to the tocentry
https://bugzilla.gnome.org/show_bug.cgi?id=753904
2015-08-21 12:20:59 +0900 Vineeth TM <vineeth.tm@samsung.com>
* ext/wavpack/gstwavpackdec.c:
wavpackdec: fix taglist memory leak
When passing the taglist to gst_audio_decoder_merge_tags, the reference is increased
by audiodecoder and the caller should free the taglist being passed.
https://bugzilla.gnome.org/show_bug.cgi?id=753903
2015-08-20 14:45:33 +0200 Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
* sys/v4l2/gstv4l2transform.c:
v4l2transform: fix pad closing
Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
https://bugzilla.gnome.org/show_bug.cgi?id=753875
2015-08-19 13:29:53 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
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* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/gst-plugins-good-plugins.signals:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-aasink.xml:
* docs/plugins/inspect/plugin-alaw.xml:
* docs/plugins/inspect/plugin-alpha.xml:
* docs/plugins/inspect/plugin-alphacolor.xml:
* docs/plugins/inspect/plugin-apetag.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-audioparsers.xml:
* docs/plugins/inspect/plugin-auparse.xml:
* docs/plugins/inspect/plugin-autodetect.xml:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-cacasink.xml:
* docs/plugins/inspect/plugin-cairo.xml:
* docs/plugins/inspect/plugin-cutter.xml:
* docs/plugins/inspect/plugin-debug.xml:
* docs/plugins/inspect/plugin-deinterlace.xml:
* docs/plugins/inspect/plugin-dtmf.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-effectv.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-flv.xml:
* docs/plugins/inspect/plugin-flxdec.xml:
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
* docs/plugins/inspect/plugin-goom.xml:
* docs/plugins/inspect/plugin-goom2k1.xml:
* docs/plugins/inspect/plugin-icydemux.xml:
* docs/plugins/inspect/plugin-id3demux.xml:
* docs/plugins/inspect/plugin-imagefreeze.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-isomp4.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mulaw.xml:
* docs/plugins/inspect/plugin-multifile.xml:
* docs/plugins/inspect/plugin-multipart.xml:
* docs/plugins/inspect/plugin-navigationtest.xml:
* docs/plugins/inspect/plugin-oss4.xml:
* docs/plugins/inspect/plugin-ossaudio.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-pulseaudio.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-rtpmanager.xml:
* docs/plugins/inspect/plugin-rtsp.xml:
* docs/plugins/inspect/plugin-shapewipe.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-smpte.xml:
* docs/plugins/inspect/plugin-soup.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speex.xml:
* docs/plugins/inspect/plugin-taglib.xml:
* docs/plugins/inspect/plugin-udp.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* docs/plugins/inspect/plugin-videobox.xml:
* docs/plugins/inspect/plugin-videocrop.xml:
* docs/plugins/inspect/plugin-videofilter.xml:
* docs/plugins/inspect/plugin-videomixer.xml:
* docs/plugins/inspect/plugin-vpx.xml:
* docs/plugins/inspect/plugin-wavenc.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
* docs/plugins/inspect/plugin-wavparse.xml:
* docs/plugins/inspect/plugin-ximagesrc.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* gst-plugins-good.doap:
* win32/common/config.h:
Release 1.5.90
2015-08-19 12:47:42 +0300 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:
Update .po files
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2015-08-19 11:29:55 +0300 Sebastian Dröge <sebastian@centricular.com>
* po/el.po:
* po/zh_CN.po:
po: Update translations
2015-08-13 17:29:58 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/multifile/gstmultifilesrc.c:
multifilesrc: fix regression with starting from index set via index property
When we haven't started yet, set the start_index when we set the index property,
so that we start at the right index position after the initial seek. The index
property was never really meant to be for writing, but it used to work, so let's
support it for backwards compatibility.
https://bugzilla.gnome.org/show_bug.cgi?id=739472
2015-08-18 10:52:11 +0100 Alex Ashley <bugzilla@ashley-family.net>
* gst/isomp4/qtdemux.c:
qtdemux: fix offset calculation when parsing CENC aux info
Commit 7d7e54ce6863ff53e188d0276d2651b65082ffdb added support for
DASH common encryption, however commit
bb336840c0b0b02fa18dc4437ce0ded3d9142801 that went onto master
shortly before the CENC commit caused the calculation of the CENC
aux info offset to be incorrect.
The base_offset was being added if present, but if the base_offset
is relative to the start of the moof, the offset was being added twice.
The correct approach is to calculate the offset from the start of the
moof and use that offset when parsing the CENC aux info.
2015-08-17 14:28:24 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/flac/gstflacenc.c:
flacenc: actually return true for accept-caps query handling
2015-08-17 14:07:10 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtp/gstrtpg723pay.c:
* gst/rtp/gstrtpgsmpay.c:
* gst/rtp/gstrtpklvpay.c:
rtp: copy metadata in the (de)payloaders which is missed before
https://bugzilla.gnome.org/show_bug.cgi?id=753706
2015-08-16 15:21:51 -0400 Dustin Spicuzza <dustin@virtualroadside.com>
* configure.ac:
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
directsoundsink: allow specifying audio playback device
https://bugzilla.gnome.org/show_bug.cgi?id=753670
2015-08-16 13:51:47 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/flac/gstflacenc.c:
flacenc: remove single entry if from loop
Iterate from the 2nd channel on and create the 1 channel struct
outside to make loop structure simpler and only slightly faster.
2015-08-16 13:21:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/flac/gstflacenc.c:
flacenc: implement proper accept-caps
Should just compare with what can be immediatelly accepted by
the element. flacenc can't renegotiate so if it has a caps already
it should only accept if it is that caps otherwise just use the
template caps
2015-08-16 13:03:36 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/flac/gstflacenc.c:
flacenc: improve sink pad template caps
Removes the need for custom caps query handling and makes it more
correct from the beginning on the template. It is a bit uglier
to read because there is 1 entry per channel but makes code easier
to maintain.
2015-08-16 12:41:56 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/y4m/gsty4mencode.c:
y4mencode: fix gst-launch version in documentation
2015-08-15 22:32:21 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/speex/gstspeexenc.c:
* ext/wavpack/gstwavpackenc.c:
* gst/law/alaw-encode.c:
* gst/law/mulaw-encode.c:
audioencoders: use template subset check for accept-caps
It is faster than doing a query that propagates downstream and
should be enough
Elements: speexenc, wavpackenc, mulawenc, alawenc
2015-08-15 22:29:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/jpeg/gstjpegenc.c:
* ext/libpng/gstpngenc.c:
* ext/vpx/gstvp8enc.c:
* ext/vpx/gstvp9enc.c:
* gst/y4m/gsty4mencode.c:
videoencoders: use template subset check for accept-caps
It is faster than doing a query that propagates downstream and
should be enough
Elements: jpegenc, pngenc, vp8enc, vp9enc, y4menc
2015-08-16 17:21:24 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/audioparsers/gstmpegaudioparse.c:
mpegaudioparse: use new baseparse API to fix tag handling
https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-03-17 17:50:37 -0400 Olivier Crête <olivier.crete@collabora.com>
* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstac3parse.c:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstdcaparse.c:
* gst/audioparsers/gstsbcparse.c:
* gst/audioparsers/gstwavpackparse.c:
audioparsers: use new base parse API to fix tag handling
https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-16 14:37:53 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/audioparsers/gstflacparse.c:
flacparse: use new baseparse API and fix tag handling
https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-16 13:04:02 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/isomp4/qtdemux.c:
qtdemux: Use signed integer type to be able to check for negative subtraction results
CID 1315829
2015-08-16 11:50:34 +0100 Luis de Bethencourt <luis@debethencourt.com>
* gst/rtp/gstrtpvorbisdepay.c:
rtpvorbisdepay: remove dead code
payload_buffer must be NULL in ignore_reserved. Check will always be false.
Introduced by b1089fb5207697ba26edb4ff66ed0f465c6df3cf
CID #1316476
2015-08-15 22:45:53 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/law/alaw-encode.c:
* gst/law/alaw-encode.h:
alawenc: port to AudioEncoder base class
2015-08-15 09:16:23 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/flac/gstflacdec.c:
* ext/speex/gstspeexdec.c:
* ext/wavpack/gstwavpackdec.c:
* gst/law/alaw-decode.c:
* gst/law/mulaw-decode.c:
audiodecoders: use default pad accept-caps handling
Avoids useless check of downstream caps when handling an
accept-caps query
Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec
2015-08-15 08:49:57 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/jpeg/gstjpegdec.c:
* ext/libpng/gstpngdec.c:
* ext/vpx/gstvp8dec.c:
* ext/vpx/gstvp9dec.c:
videodecoders: use default pad accept-caps handling
Avoids useless check of downstream caps when handling an
accept-caps query
Elements: jpegdec, pngdec, vp8dec, vp9dec
2015-08-15 11:31:04 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/law/alaw-decode.c:
alawdec: make error handling a bit nicer
Print the element along with the debug to make it easier to trace
the failures
2015-08-15 11:04:16 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/law/alaw-decode.c:
* gst/law/alaw-decode.h:
alawdec: port to audiodecoder base class
mulawdec was already ported, alawdec was left behind.
2015-08-15 10:34:14 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/isomp4/qtdemux.c:
qtdemux: only look for more samples in moofs in pull-mode
For playback of some fragmented formats with qtdemux it will
try to look for the next moof after finishing one but it is only
possible for pull-mode. For playback of streaming fragmented formats
such as DASH it should just not try to look for another moof but
instead wait for more data.
https://bugzilla.gnome.org/show_bug.cgi?id=752602
https://bugzilla.gnome.org/show_bug.cgi?id=752603
2015-08-15 12:58:50 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/audioparsers/gstdcaparse.c:
dcaparse: Don't look for a second syncword
There are streams out there that consistently contain garbage between
every frame so we never ever find a second consecutive syncword.
See https://bugzilla.gnome.org/show_bug.cgi?id=738237
2015-08-15 11:12:05 +0100 Tim-Philipp Müller <tim@centricular.com>
* ext/vpx/gstvp8enc.c:
* ext/vpx/gstvp9enc.c:
vp8enc, vp9enc: reset multipass file index when stopping encoder
Fixes multipass encoding when re-using the same element/pipeline
for subsequent encoding runs.
https://bugzilla.gnome.org/show_bug.cgi?id=747728
2015-08-15 11:09:42 +0100 Tim-Philipp Müller <tim@centricular.com>
* ext/vpx/gstvp9enc.c:
* ext/vpx/gstvp9enc.h:
vp9enc: provide support for multiple pass cache files
Some files may provide different caps insight of one stream. Since
vp9enc support caps reinit, we should support cache reinit too.
If more then file cache file will be created, the naming will be:
cache cache.1 cache.2 ...
Based on patch by: Oleksij Rempel <linux@rempel-privat.de>
https://bugzilla.gnome.org/show_bug.cgi?id=747728
2015-08-14 11:41:42 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* tests/check/elements/aacparse.c:
tests: aacparse: use caps query instead of accept-caps
The accept-caps query just does a shallow check at the current
element while at this test we want it to also look at downstream.
So use caps query there.
https://bugzilla.gnome.org/show_bug.cgi?id=753623
2015-08-14 11:40:22 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstac3parse.c:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstdcaparse.c:
* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstmpegaudioparse.c:
* gst/audioparsers/gstsbcparse.c:
* gst/audioparsers/gstwavpackparse.c:
audioparsers: enable accept-template flag
Do a quick check with the pad template caps as it is enough. Users
should have figured the appropriate full caps on a previous caps query
https://bugzilla.gnome.org/show_bug.cgi?id=753623
2015-08-14 15:46:53 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
rtspsrc: send the User-Agent header
Sometimes it is useful to know this information on the
server side. Other popular implementations (vlc, ffmpeg, ...)
also send this header on every message.
This includes a new "user-agent" property that the user
can set to use a custom User-Agent string. The default
is "GStreamer/<version>"
https://bugzilla.gnome.org/show_bug.cgi?id=750101
2015-08-14 15:42:42 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: wrap gst_rtsp_message_init_request in a local function
This will allow adding common request initialization, like the
user agent string, in just one place.
2015-08-14 09:36:09 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
* gst/audiofx/audioecho.c:
audioecho: make sure buffer gets reallocated if max_delay changes
https://bugzilla.gnome.org/show_bug.cgi?id=753490
2015-07-09 09:51:26 +0200 Oleksij Rempel <linux@rempel-privat.de>
* ext/vpx/gstvp8enc.c:
* ext/vpx/gstvp8enc.h:
vp8enc: provide support for multiple pass cache files
Some files may provide different caps insight of one stream. Since vp8enc
support caps reinit, we should support cache reinit too.
If more then file cache file will be created, the naming will be:
cache
cache.1
cache.2
...
https://bugzilla.gnome.org/show_bug.cgi?id=747728
2015-04-15 22:51:51 +0200 Ramiro Polla <ramiro.polla@collabora.co.uk>
* gst/rtp/gstrtpmp4gdepay.c:
rtpmp4gdepay: fix timestamps for RTP packets with multiple AUs
Use constantDuration to calculate the timestamp of non-first AU in the
RTP packet.
If constantDuration is not present in the MIME parameters, its value
must be calculated based on the timing information from two consecutive
RTP packets with AU-Index equal to 0.
https://bugzilla.gnome.org/show_bug.cgi?id=747881
2015-08-14 06:43:13 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: remove unnecessary if, g_free is null safe
2015-08-14 08:33:56 +0100 Alex Ashley <bugzilla@ashley-family.net>
* ext/soup/gstsouphttpsrc.c:
* ext/soup/gstsouphttpsrc.h:
souphttpsrc: add property to set HTTP method
To allow souphttpsrc to be use HTTP methods other than GET
(e.g. HEAD), add a "method" property that is a string. If this
property is not set, GET is used.
https://bugzilla.gnome.org/show_bug.cgi?id=752413
2015-08-14 11:13:01 +0200 Edward Hervey <bilboed@bilboed.com>
* tests/check/generic/states.c:
check: Rename states unit test
Makes it easier to differentiate from other modules states unit test
2015-08-14 09:21:25 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/goom/gstaudiovisualizer.c:
* gst/goom/gstaudiovisualizer.h:
* gst/goom2k1/gstaudiovisualizer.c:
* gst/goom2k1/gstaudiovisualizer.h:
goom: Rename get_type() function of base class to prevent symbol conflicts
This is a problem when statically linking.
2015-08-13 16:32:55 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: Keep the DTS estimate if we got no DTS after a jitterbuffer reset
Otherwise we will just output buffers without timestamps after a reset if no
timestamps are provided by upstream, e.g. when using RTSP over TCP.
https://bugzilla.gnome.org/show_bug.cgi?id=749536
2015-08-12 17:16:01 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
* gst/matroska/matroska-demux.h:
* gst/matroska/matroska-parse.h:
matroska: Remove unused variable
https://bugzilla.gnome.org/show_bug.cgi?id=753556
2015-08-04 20:59:17 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpL24depay.c:
* gst/rtp/gstrtpac3depay.c:
* gst/rtp/gstrtpac3pay.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpbvdepay.c:
* gst/rtp/gstrtpceltdepay.c:
* gst/rtp/gstrtpceltpay.c:
* gst/rtp/gstrtpdvdepay.c:
* gst/rtp/gstrtpdvpay.c:
* gst/rtp/gstrtpg722depay.c:
* gst/rtp/gstrtpg723pay.c:
* gst/rtp/gstrtpg726depay.c:
* gst/rtp/gstrtpg729depay.c:
* gst/rtp/gstrtpg729pay.c:
* gst/rtp/gstrtpgsmdepay.c:
* gst/rtp/gstrtpgsmpay.c:
* gst/rtp/gstrtpgstdepay.c:
* gst/rtp/gstrtpgstpay.c:
* gst/rtp/gstrtph261depay.c:
* gst/rtp/gstrtph261pay.c:
* gst/rtp/gstrtph263depay.c:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtph264depay.c:
* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpj2kdepay.c:
* gst/rtp/gstrtpj2kpay.c:
* gst/rtp/gstrtpjpegdepay.c:
* gst/rtp/gstrtpjpegpay.c:
* gst/rtp/gstrtpmp1sdepay.c:
* gst/rtp/gstrtpmp2tdepay.c:
* gst/rtp/gstrtpmp2tpay.c:
* gst/rtp/gstrtpmp4adepay.c:
* gst/rtp/gstrtpmp4apay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vdepay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtpmpvdepay.c:
* gst/rtp/gstrtpmpvpay.c:
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtpqcelpdepay.c:
* gst/rtp/gstrtpqdmdepay.c:
* gst/rtp/gstrtpsbcdepay.c:
* gst/rtp/gstrtpsbcpay.c:
* gst/rtp/gstrtpsirendepay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtp/gstrtpsv3vdepay.c:
* gst/rtp/gstrtptheoradepay.c:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtptheorapay.h:
* gst/rtp/gstrtputils.c:
* gst/rtp/gstrtputils.h:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbispay.c:
* gst/rtp/gstrtpvorbispay.h:
* gst/rtp/gstrtpvp8depay.c:
* gst/rtp/gstrtpvp8pay.c:
* gst/rtp/gstrtpvrawdepay.c:
* gst/rtp/gstrtpvrawpay.c:
rtp: Copy metadata in the (de)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
2015-08-10 18:20:15 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/isomp4/qtdemux.c:
qtdemux: fix small typo in comment
2015-08-10 16:19:18 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/goom2k1/gstgoom.c:
goom2k1/doc: Fixup previous commit
2015-08-10 15:55:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/goom2k1/gstgoom.c:
* gst/goom2k1/gstgoom.h:
goom2k1/doc: Use GstGoom2k1 namespace
The doc generator isn't happy when we have class name clash. Simply
use it's own namespace.