ChangeLog
=========
2017-04-27 Sebastian Dröge <[email protected]>
* configure.ac:
releasing 1.11.91
2017-04-27 15:28:02 +0300 Sebastian Dröge <[email protected]>
* po/LINGUAS:
* po/el.po:
* po/fur.po:
po: Update translations
2017-04-27 12:56:27 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Don't crash in debug output if stream==NULL
That case is correctly handled below but not in the debug output.
https://bugzilla.gnome.org/show_bug.cgi?id=781270
2017-04-25 17:11:27 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Don't perform seeks with inconsistent seek values
If gst_segment_do_seek() fails, we shouldn't try seeking on that
resulting segment but just error out. Crashes further down the line
otherwise.
2017-04-24 20:27:49 +0100 Tim-Philipp Müller <[email protected]>
* common:
Automatic update of common submodule
From 60aeef6 to 48a5d85
2017-04-24 17:31:04 +0100 Tim-Philipp Müller <[email protected]>
* tests/check/Makefile.am:
* tests/check/elements/rtp-payloading.c:
tests: rtp-payloading: add test for rtph264depay avc/byte-stream
output
Make sure avc output doesn't contain SPS/PPS inline, but
byte-stream output does.
2017-04-24 17:29:37 +0100 Tim-Philipp Müller <[email protected]>
* gst/rtp/gstrtph264depay.c:
rtph264depay: don't insert SPS/PPS inline for AVC output
SPS/PPS are in the caps in this case and shouldn't be in
the stream data.
2017-04-21 19:09:14 +0100 Sebastian Dröge <[email protected]>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Chain up to the parent class' provide_clock() implementation
If no clock was provided directly by rtspsrc. This behaviour was
removed
by f8013487c91a6ffc552a4b25aa1a70f0bd5377f8 and results in rtspsrc not
providing the system clock via the rtpjitterbuffer.
As a result, if another element like an audio sink, provides a clock,
the pipeline would select that (when going to PAUSED/PLAYING again
later).
Audio clocks usually don't progress in PAUSED, and thus our live
source
won't be able to use the clock to produce data, making the sink never
preroll and everything is stuck.
2017-04-20 11:22:15 +0200 Jürgen Sachs <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: reset sample_description_id to default
Fixes stream where sample_description_id is specified in the tfhd
https://bugzilla.gnome.org/show_bug.cgi?id=778337
2017-04-20 13:16:24 +0100 Sebastian Dröge <[email protected]>
* gst/multifile/gstsplitmuxsink.c:
splitmuxsink: Don't use an explicit name for requesting audio pads
... unless the muxer uses the same audio pad template name as
splitmuxsink. We can't request a pad called "audio_0" on a muxer that
wants pads to be "sink_%d".
2017-02-23 09:31:36 +0900 ChangBok Chae <[email protected]>
* gst/flv/gstflvdemux.c:
flvdemux: remove duplicated segment initialization
It's also done in gst_flv_demux_cleanup().
https://bugzilla.gnome.org/show_bug.cgi?id=779106
2017-04-20 20:17:35 +1000 Xavier Claessens <[email protected]>
* gst/multifile/gstsplitmuxsink.c:
splitmuxsink: Correctly catch FLUSH events in probes
https://bugzilla.gnome.org/show_bug.cgi?id=767498
2017-04-19 12:28:12 +0100 Tim-Philipp Müller <[email protected]>
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
Revert "rtpbin: pipeline gets an EOS when any rtpsources byes"
This reverts commit eeea2a7fe88a17b15318d5b6ae6e190b2f777030.
It breaks EOS in some sender pipelines, see
https://bugzilla.gnome.org/show_bug.cgi?id=773218#c20
2017-04-14 17:01:49 +0200 Edward Hervey <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Reset adapter in more discontinuity cases
In push mode we process as much as possible in the adapter. When we
receive
a DISCONT buffer which we can't match to an actual sample (based on
the existing
sample table) and there is still data remaining in the incoming
adapter,there is
one of two cases happening:
1) We are doing reverse playback, in which case we should flush out
all pending
data
2) We have leftover data from the previous incoming buffer... which
we can't do
anything about.
For the second case, make sure we flush out the remaining data so
that we can start
parsing again from scratch.
https://bugzilla.gnome.org/show_bug.cgi?id=781319
2017-04-14 10:56:41 +0200 Edward Hervey <[email protected]>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Use GST_ELEMENT_ERROR_WITH_DETAILS
Allows the application to know the exact status code that was returned
by the server in a programmatic fashion.
https://bugzilla.gnome.org/show_bug.cgi?id=781304
2017-04-16 18:47:56 +0900 Seungha Yang <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Fix leak on QtDemuxStreamStsdEntry
Fix unit test failure
https://bugzilla.gnome.org/show_bug.cgi?id=781362
2017-04-14 13:38:53 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/atoms.c:
* gst/isomp4/atoms.h:
* gst/isomp4/gstqtmux.c:
qtmux: Fix timescale of timecode tracks
They should have ideally the same timescale of the video track, which
we
can't guarantee here as in theory timecode configuration and video
framerate could be different. However we should set a correct
timescale
based on the framerate given in the timecode configuration, and not
just
use the framerate numerator.
2017-04-13 13:25:06 +0200 Edward Hervey <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Properly reset demuxer when all streams are EOS
Make sure offset and neededbytes are properly resetted when all
streams are EOS in push-mode.
Avoids cases when some data might still be pushed by upstream (because
it didn't yet see the resulting GST_FLOW_EOS yet) and qtdemux gets
completely lost.
https://bugzilla.gnome.org/show_bug.cgi?id=781266
2017-04-13 08:00:30 +0200 Edward Hervey <[email protected]>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: Make more usage of error macro
And make sure we actually use the provided soup_msg argument in the
macro
2017-04-12 18:46:53 +0530 Nirbheek Chauhan <[email protected]>
* ext/meson.build:
meson: Print message when disabling taglib on MSVC
2017-04-12 13:26:59 +0200 Edward Hervey <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: Don't forget to update pad->last_buf
buf is the current pad->last_buf value. If ever it gets
copied/unreffed,
we need to make sure to write back the new pointer to the last_buf
variable.
Fixes using wrong pointer values in the case of decrasing DTS value
2017-04-12 11:33:05 +0200 Edward Hervey <[email protected]>
* tests/check/elements/.gitignore:
tests: Add vp9enc to gitignore
2017-04-11 13:41:48 +0200 Jürgen Sachs <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: fix: sample description index override in tfhd not evaluated
https://bugzilla.gnome.org/show_bug.cgi?id=778337
2017-04-12 11:03:24 +0200 Edward Hervey <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Add out-of-bound check
Make sure we don't read invalid memory
2016-04-27 12:17:37 -0300 Thiago Santos <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: move parsing of tkhd out of stsd entry loop
It needs only to be read once.
2016-04-07 12:23:35 -0300 Thiago Santos <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: check for a different stsd entry before pushing a sample
Before pushing a sample, check if there was a change in the current
stsd entry. This patch also assumes that the first stsd entry is
used as default for the first sample. It might cause an uneeded
caps renegotiation when this isn't the case.
2016-04-06 12:55:18 -0300 Thiago Santos <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: parse all stsd entries
stsd can have multiple format entries, parse them all.
This is required to play DVB DASH profile that uses multiple entries
to identify the different available bitrates/options on dash streams
The stream format-specific data is not stored into
QtDemuxStreamStsdEntry
2016-04-05 14:34:00 -0300 Thiago Santos <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: rework stsd sample entries access
Instead of using the stsd as a base pointer, use the actual stsd
entry as the stsd can have multiple entries. This is rarely used
for file playback but is a possible profile with in DVB DASH specs.
This still doesn't support stsd with multiple entries but makes it
easier to do so.
2016-04-05 18:00:10 -0300 Thiago Santos <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: get stsd child by index instead of type
There might be multiple children with the same type
2017-04-07 16:33:18 +0300 George Kiagiadakis <[email protected]>
* tests/check/elements/rtprtx.c:
tests/check/rtprtx: add checks for rtprtxqueue's
max-size-{time,packets} properties
https://bugzilla.gnome.org/show_bug.cgi?id=780867
2017-04-04 17:33:31 +0300 George Kiagiadakis <[email protected]>
* gst/rtpmanager/gstrtprtxqueue.c:
* gst/rtpmanager/gstrtprtxqueue.h:
rtprtxqueue: implement handling of the max-size-time property
https://bugzilla.gnome.org/show_bug.cgi?id=780867
2017-04-10 23:49:06 +0100 Tim-Philipp Müller <[email protected]>
* autogen.sh:
* common:
Automatic update of common submodule
From 39ac2f5 to 60aeef6
2017-04-10 08:56:00 +0000 Todor Tomov <[email protected]>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2object: Copy timestamp when importing buffers
This is needed for V4L2_OUTPUT interface, and is harmless of
V4L2_CAPTURE interfaces. This will fix timestamp in cases like:
v4l2src io-mode=dmabuf ! v4l2videoNenc output-io-mode=dmabuf-import !
...
Same apply for userptr.
https://bugzilla.gnome.org/show_bug.cgi?id=781119
2017-04-10 15:55:30 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: Fix last_dts tracking for raw audio and similar formats
Accumulate the durations directly and don't scale yet another time by
the number of samples.
2017-04-07 10:48:50 +0100 Vincent Penquerc'h
<[email protected]>
* tests/check/elements/splitmux.c:
tests: fix leak in splitmux test
https://bugzilla.gnome.org/show_bug.cgi?id=781025
2017-04-07 15:29:43 +0800 Lyon Wang <[email protected]>
* gst/audiofx/gstscaletempo.c:
scaletempo: Scale GAP event timestamp and duration like for buffers
https://bugzilla.gnome.org/show_bug.cgi?id=781008
2017-02-17 10:01:08 -0300 Thibault Saunier <[email protected]>
* sys/v4l2/gstv4l2videodec.c:
* sys/v4l2/gstv4l2videodec.h:
v4l2dec: Fix race when going from PAUSED to READY
Running `gst-validate-launcher -t
validate.file.playback.change_state_intensive.vorbis_vp8_1_webm`
on odroid XU4 (s5p-mfc v4l2 driver) often leads to:
ERROR:../subprojects/gst-plugins-good/sys/v4l2/gstv4l2videodec.c:215:gst_v4l2_video_dec_stop:
assertion failed: (g_atomic_int_get (&self->processing) == FALSE)
This happens when the following race happens:
- T0: Main thread
- T1: Upstream streaming thread
- T2. v4l2dec processing thread)
[The decoder is in PAUSED state]
T0. The validate scenario runs `Executing (36/40) set-state:
state=null repeat=40`
T1- The decoder handles a frame
T2- A decoded frame is push downstream
T2- Downstream returns FLUSHING as it is already flushing changing
state
T2- The decoder stops its processing thread and sets `->processing =
FALSE`
T1- The decoder handles another frame
T1- `->process` is FALSE so the decoder restarts its streaming thread
T0- In v4l2dec-> stop the processing thread is stopped
NOTE: At this point the processing thread loop never started.
T0- assertion failed: (g_atomic_int_get (&self->processing) == FALSE)
Here I am removing the whole ->processing logic to base it all on the
GstTask state to avoid duplicating the knowledge.
https://bugzilla.gnome.org/show_bug.cgi?id=778830
=== release 1.11.90 ===
2017-04-07 16:31:56 +0300 Sebastian Dröge <[email protected]>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-good-plugins.args:
* 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-shout2.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:
* meson.build:
Release 1.11.90
2017-04-07 15:18:11 +0300 Sebastian Dröge <[email protected]>
* 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
2017-04-07 15:06:30 +0300 Sebastian Dröge <[email protected]>
* po/el.po:
po: Update translations
2017-04-06 12:01:00 +0200 Edward Hervey <[email protected]>
* gst/audioparsers/gstaacparse.c:
aacparse: streamline and improve AudioSpecificConfig parsing
AudioSpecifigConfig is used in a variety of AAC streams but was
being parsed differently. Instead, make everyone use the same parsing.
* Remove unused 'bits' field (it was always set to 0 if present)
* Add proper GAConfig parsing (to know the number of samples per
frame
if present).
Fixes wrong rate/channels configuration in streams coming from qtdemux
https://bugzilla.gnome.org/show_bug.cgi?id=780966
2017-04-05 09:46:31 -0400 Nicolas Dufresne <[email protected]>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: Fix 32bit only printf format
The previous patch was using %llu for 64bits printf, which is 32bit
specific. We also trace the latency in time human readable form now.
2016-03-16 16:22:48 +0100 Philipp Zabel <[email protected]>
* sys/v4l2/gstv4l2object.c:
v4l2object: set streamparm for outputs that support it
Without a specified framerate from the sink, the decoder frame
interval
should be set using the framerate of the encoded video stream.
Therefore, the v4l2object should be able to change the framerate on
the
output if the V4L2 device accepts it.
This is also necessary for mem2mem encoders so that their bitrate
calculation code may work correctly and they may report the correct
frame duration on the capture queue.
https://bugzilla.gnome.org/show_bug.cgi?id=779466
2016-03-16 16:24:55 +0100 Philipp Zabel <[email protected]>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: only set latency if the frame duration is valid
If the duration of the v4l2object is GST_CLOCK_TIME_NONE, because the
sink did not specify a framerate in the caps and the driver accepts
the
framerate, the decoder element uses GST_CLOCK_TIME_NONE to calculate
and
set the element latency.
While this is a bug of the capture driver, the decoder element should
not use the invalid duration to calculate a latency, but print a
warning
instead.
https://bugzilla.gnome.org/show_bug.cgi?id=779466
2016-11-23 12:17:55 -0500 Olivier Crête <[email protected]>
* sys/v4l2/gstv4l2sink.c:
v4l2sink: Block in preroll_wait on unlock
The correct behaviour of anything stuck in the ->render() function
between ->unlock() and ->unlock_stop() is to call
gst_base_sink_wait_preroll() and only return an error if this returns
an
error, otherwise, it must continue where it left off!
https://bugzilla.gnome.org/show_bug.cgi?id=774945
2017-04-05 15:55:20 +1000 Jan Schmidt <[email protected]>
* ext/vpx/gstvp9dec.c:
vp9dec: Add warnings for unsupported frame formats
At least output an element warning on the bus when we
encounter a frame format GStreamer doesn't currently support.
2017-04-04 17:55:13 +0200 Edward Hervey <[email protected]>
* gst/audioparsers/gstaacparse.c:
aacparse: Handle Parametric Stereo with HE-AAC(v2)
According to ISO/IEC:14496-2:2009 , in the case of HE-AACv2
(audioObjecType
29) parametric stereo is used (a single mono track is used and then
transformations are applied to it to provide a stereo output).
We therefore report two channels in the case where there is one
reported
in the audioChannelConfiguration.
Fixes the various issues where a demuxer would report two channels,
but
then the parser would say there's only one channel, and then the
decoder
would output two channels.
2017-04-04 15:22:25 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: Simplify buffer refcounting in add_buffer() and remove
unneeded NULL checks
2017-04-04 15:08:33 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: Select the best pad based on the cached last_buf if any
last_buf is the one we're going to write next, not buf. As such we
should check timestamps against that one if there is one to select the
earliest pad.
Also remember the currently selected pad in the very beginning when
storing the first last_buf.
This both solves some edge cases where not the correct next pad was
selected corresponding to the target interleave.
2017-04-04 15:07:40 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: Error out immediately if a timecode is to be written but
downstream return not-OK
2017-04-03 11:34:49 +0200 Edward Hervey <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Update variables before early exit
This is an update of d78d5896272d78df41e696fac929e7dfb3bb3dfa
We still exit as early as possible in case of non-ok/non-unlinked
combined
flow, but we first make sure that we update the internal position
variables.
This ensures that if upstreams "ignores" the flow return (and carries
on pushing),
we don't end up processing data with completely bogus
variables/positions.
2017-03-24 00:11:13 +1300 Douglas Bagnall <[email protected]>
* gst/interleave/interleave.c:
* gst/interleave/interleave.h:
interleave: avoid using uninitialised ordering_map
If self->channel_positions == NULL (which seems unlikely),
self->default_channels_ordering_map will be used unintialised.
We avoid that by keeping track of the channel_mask, which is set when
the ordering map is initialised.
https://bugzilla.gnome.org/show_bug.cgi?id=780331
2017-03-23 23:56:31 +1300 Douglas Bagnall <[email protected]>
* gst/interleave/interleave.c:
interleave: don't overflow channel map with >64 channels
When there are more than 64 channels, we don't want to exceed the
bounds of the ordering_map buffer, and in these cases we don't want to
rempa at all. Here we avoid doing that.
https://bugzilla.gnome.org/show_bug.cgi?id=780331
2017-03-28 14:23:16 -0300 Thibault Saunier <[email protected]>
* tests/check/meson.build:
meson: Use get_pkgconfig_variable instead of calling pkg-config
ourself
It is avalaible in meson 0.36 which is now are requirement
2017-03-28 14:22:41 -0300 Thibault Saunier <[email protected]>
* pkgconfig/gstreamer-plugins-good.pc.in:
* pkgconfig/meson.build:
pkgconfig: Do not ever build an installed .pc file
2017-03-28 11:15:53 -0300 Thibault Saunier <[email protected]>
* tests/check/meson.build:
meson: test: Fix environment object usage
2017-03-28 11:14:47 -0300 Thibault Saunier <[email protected]>
* meson.build:
* pkgconfig/gstreamer-plugins-good.pc.in:
* pkgconfig/meson.build:
pkgconfig: Generate the pkg-config with meson too
2017-03-27 21:52:00 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: In gap mode, consider the mdat offset when calculating the
remaining mdat size
The mdat generally does not start at offset 0, we have to include the
size of the moof and whatever else was in front of the mdat.
2017-03-27 11:43:31 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/atomsrecovery.c:
atomsrecovery: Error out when fseek() fails instead of silently
ignoring
CID 1403262
2017-03-23 22:13:05 +0100 Carlos Rafael Giani <[email protected]>
* sys/v4l2/gstv4l2object.c:
v4l2object: Also add videometa if there is padding to the right and
bottom
https://bugzilla.gnome.org/show_bug.cgi?id=780478
2017-03-21 12:54:27 +0200 George Kiagiadakis <[email protected]>
* gst/rtpmanager/gstrtpmux.c:
rtpmux: fix output segment and buffer DTS to correspond to the
flattened PTS
https://bugzilla.gnome.org/show_bug.cgi?id=780347
2017-03-23 17:53:19 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/gstqtmux.c:
* gst/isomp4/gstqtmux.h:
qtmux: Remove some unused variables
2017-03-23 15:01:16 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: Remove a couple of unneeded levels of indentation
2017-03-22 18:18:40 +0000 Enrique Ocaña González <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: distinguish TFDT with value 0 from no TFDT at all
TFDTs with time 0 are being ignored since commit 1fc3d42f. They're
mistaken with the case of not having TFDT, but those two cases
must be distinguished in some way.
This patch passes an extra boolean flag when the TFDT is present.
This is now the condition being evaluated, instead of checking for
0 time.
https://bugzilla.gnome.org/show_bug.cgi?id=780410
2017-03-22 19:15:09 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: Reset current chunk after writing out timecode
If we have multiple tracks with timecodes, or it's not the first track
that has timecodes, or not the first buffer, we already started a
chunk
for media data. We now need to "close" that chunk because we wrote
data
for the timecode track and a new chunk has to be started for the
original track the next time it has data.
2017-03-22 18:52:51 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/gstqtmux.c:
* gst/isomp4/gstqtmux.h:
qtmux: Do timecode handling per track, not per muxer instance
There could be multiple video tracks with timecodes.
2017-03-22 00:38:51 +1100 Jan Schmidt <[email protected]>
* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
qtdemux: matroskademux: Ignore repeated seek events
Similar to what was done in adaptivedemux, ignore seek
events we've already handled - such as when they are received
on every srcpad of files with lots of streams.
2017-03-21 14:55:32 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
dashdemux: Update mdatleft from overall mdatsize and offset when
observing a gap
Otherwise mdatleft will have a value calculated from the initial
mdatsize minus the parts of the stream that we saw, which is not
including all the parts of the stream that might've been skipped.
2017-03-20 17:03:32 +0000 Tim-Philipp Müller <[email protected]>
* ext/soup/gstsouphttpsrc.c:
* gst/audioparsers/gstmpegaudioparse.c:
docs: update two references to the removed 'mad' plugin
https://bugzilla.gnome.org/show_bug.cgi?id=776140
2017-03-20 12:03:29 +0200 George Kiagiadakis <[email protected]>
* gst/rtpmanager/gstrtprtxqueue.c:
rtprtxqueue: add basic documentation and example pipelines
Mostly explaining the difference between rtprtxqueue and rtprtxsend.
2017-03-17 20:58:28 -0400 Nicolas Dufresne <[email protected]>
* sys/v4l2/meson.build:
v4l2: Fix meson plugin shared object name
It didn't match between AutoMake and Meson, and the Meson name
didn't math the plugin name (video4linux2).
2017-03-16 18:20:54 +0200 George Kiagiadakis <[email protected]>
* gst/rtpmanager/gstrtprtxreceive.c:
rtprtxreceive: fix example pipelines and improve the documentation
https://bugzilla.gnome.org/show_bug.cgi?id=771383
2017-03-17 14:10:40 +0000 Vincent Penquerc'h
<[email protected]>
* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstflacparse.h:
flacparse: fix playback if sample number does not start at 0
This reverts commit 29b807685d3c962bbe8afe351c5dca97d59eb5e0, while
fixing the original breaking tests/check/pipelines/flacdec.
2017-03-17 11:30:04 +0000 Vincent Penquerc'h
<[email protected]>
* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstflacparse.h:
Revert "flacparse: fix playback if sample number does not start at 0"
This breaks gst-validate on the build server (though not locally),
and a unit test, and I can't run unit tests right now for some
unrelated reason.
This reverts commit 0747b56f8e7f4731d67f8d13a4bdc453dde0fdf7.
2017-03-16 17:44:41 +0200 George Kiagiadakis <[email protected]>
* gst/rtpmanager/rtpsession.c:
rtpsession: print the correct variable in debug statement
This debug statement is meant to print the time since the last (early)
RTCP transmission, not the last regular RTCP transmission (which also
happens to be set a few lines above to current_time, so the debug
output
is just confusing)
2017-03-16 17:42:27 +0200 George Kiagiadakis <[email protected]>
* gst/rtpmanager/gstrtprtxsend.c:
rtprtxsend: convert LOG message to TRACE
This is printed too often (for every chained buffer!) and just
clutters the logs.
2017-03-16 14:58:45 +0100 Miguel París Díaz <[email protected]>
* gst/rtpmanager/rtpsource.c:
rtpsource: fix warning message
https://bugzilla.gnome.org/show_bug.cgi?id=780105
2017-03-16 13:54:54 +0000 Vincent Penquerc'h
<[email protected]>
* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstflacparse.h:
flacparse: fix playback if sample number does not start at 0
https://bugzilla.gnome.org/show_bug.cgi?id=777738
2017-03-15 18:58:55 +0100 Miguel París Díaz <[email protected]>
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
rtpsource: get clock-rate from pt if needed to generate SR
https://bugzilla.gnome.org/show_bug.cgi?id=780105
2017-03-16 13:52:48 +0200 Sebastian Dröge <[email protected]>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: Include GStreamer souphttpsrc version in default
User-Agent string
2017-03-16 00:41:44 +0000 Tim-Philipp Müller <[email protected]>
* gst/rtp/gstrtph264depay.c:
rtph264depay: fix crash with empty sprops-parameters
https://bugzilla.gnome.org/show_bug.cgi?id=780040
2017-03-11 21:20:40 -0800 Thiago Santos <[email protected]>
* gst/isomp4/atomsrecovery.c:
* gst/isomp4/atomsrecovery.h:
atomsrecovery: also handle extra atoms after 'mdia' in a 'trak'
Take into account the atoms at the end of the 'trak' atom when
recovering it. So that its size (already computed and added in the
trak
size) isn't making offsets wrong.
https://bugzilla.gnome.org/show_bug.cgi?id=771478
2017-03-11 12:56:33 -0800 Thiago Santos <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: avoid fallthrough to moovrecovery failure section
Return before that to preserve our successfull results, otherwise no
moov recovery information would be written
https://bugzilla.gnome.org/show_bug.cgi?id=771478
2017-03-11 12:27:28 -0800 Thiago Santos <[email protected]>
* gst/isomp4/atomsrecovery.c:
atomsrecovery: expect more atom types at the headers
Skip more atoms at the header until it finds the 'mdat' to continue
the
moov recovery
https://bugzilla.gnome.org/show_bug.cgi?id=771478
2017-03-14 16:42:25 -0400 Olivier Crête <[email protected]>
* Makefile.am:
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/pulse/.gitignore:
* tests/examples/pulse/Makefile.am:
* tests/examples/pulse/pulse.c:
pulse example: Remove
That example only tested the property probe interface, which has been
removed.
The same kind of thing can now be done with the generic
gst-device-monitor tool.
2017-03-14 16:38:02 -0400 Olivier Crête <[email protected]>
* sys/v4l2/gstv4l2object.h:
v4l2: Remove unused macro
2017-03-14 16:35:25 -0400 Olivier Crête <[email protected]>
* sys/v4l2/gstv4l2object.c:
* sys/v4l2/gstv4l2object.h:
v4l2: Remove unused definitions
2017-03-14 10:10:19 +0100 Emeric Grange <[email protected]>
* gst/isomp4/fourcc.h:
* gst/isomp4/gstqtmux.c:
* gst/isomp4/gstqtmuxmap.c:
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux_types.c:
qtmux: add CineForm support
https://bugzilla.gnome.org/show_bug.cgi?id=780024
2017-03-14 15:09:44 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: Only create new chunks if we have more than a single stream
There's no point in creating multiple chunks otherwise, it only wastes
some bytes for storing the chunk offsets.
2017-03-14 10:09:46 +0100 Emeric Grange <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: add S16L support
https://bugzilla.gnome.org/show_bug.cgi?id=780022
2017-03-14 15:48:08 +1100 Jan Schmidt <[email protected]>
* tests/check/elements/splitmux.c:
splitmux test: Use passed first/last timestamps
Don't hard-code the expected timestamp range, use the
values the caller is passing in.
2017-03-12 11:42:25 -0400 Nicolas Dufresne <[email protected]>
* Makefile.am:
* docs/plugins/inspect/plugin-soup.xml:
Add old plugin names to cruft list
This will help fixing uninstalled setup. Also fix missing path
correction in one of the plugin xml.
https://bugzilla.gnome.org/show_bug.cgi?id=779344
2016-12-15 12:38:40 +0100 Michael Dutka <[email protected]>
* gst/rtp/gstrtph264depay.c:
* gst/rtp/gstrtph265depay.c:
rtph264depay, rtph265depay: remove stray g_debug()
https://bugzilla.gnome.org/show_bug.cgi?id=779858
2017-03-10 11:24:14 +0100 Wim Taymans <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: init fourcc
Initialize the fourcc to 0 so that we can detect failure later.
2017-03-08 22:50:52 -0500 Nicolas Dufresne <[email protected]>
* tests/check/Makefile.am:
* tests/check/elements/level.c:
* tests/check/elements/rglimiter.c:
tests: Add missing LDADD for libm in tests using math.h
Also, remove the math.h include for the one that just prentend to need
it.
2017-03-08 22:15:46 -0500 Nicolas Dufresne <[email protected]>
* Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
Fix shout2 plugin doc generation
In the previous patch, we also renamed shout2send to shout2, so it
does
not clash with it's feature. Though we forgot to rename it in the doc
reference. This patch also add a cruft detection on the xml that made
me
miss this error.
https://bugzilla.gnome.org/show_bug.cgi?id=779344
2017-03-04 10:52:47 -0500 Nicolas Dufresne <[email protected]>
* docs/plugins/inspect/plugin-oss4.xml:
* docs/plugins/inspect/plugin-pulseaudio.xml:
* docs/plugins/inspect/plugin-shout2.xml:
* ext/pulse/Makefile.am:
* ext/pulse/meson.build:
* ext/shout2/gstshout2.c:
* ext/soup/Makefile.am:
* ext/soup/meson.build:
* sys/oss4/Makefile.am:
Fix plugin filenames to match plugin names
- libgstpulse.so becomes libgstpulseaudio.so
- libgstsouphttpsrc.so becomes libgstsoup.so
- libgstoss4audio.so becomes libgstoss4.so
https://bugzilla.gnome.org/show_bug.cgi?id=779344
2017-03-08 16:01:02 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/atoms.c:
qtmux: Free EDTS instead of just clearing it and setting it to NULL
2017-03-08 15:27:32 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/atoms.c:
* gst/isomp4/gstqtmux.c:
qtmux: Fix some memory leaks related to timecode tracks
2017-03-04 00:34:44 +1100 Jan Schmidt <[email protected]>
* tests/check/elements/splitmux.c:
splitmux: Add unit test for reverse playback
Ensure that reverse playback works and generates the range
of timestamps (0-3s) we expect, in monotonically descending order.
2017-02-28 11:50:45 +1100 Jan Schmidt <[email protected]>
* gst/multifile/gstsplitmuxsrc.c:
splitmuxsrc: Fix reverse playback
Fix the check for whether the start time of the segment has
been reached when playing in reverse. Otherwise, playback
stops after reaching the start of any file part, instead of
continuing until all parts within the segment have played
2017-02-22 03:01:31 +1100 Jan Schmidt <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Don't lose crypto info on a new moof
We parse the next moof in advance of having pushed
all samples from the previous one in some cases, and
we'll still need the crypto info from the previous
fragment so keep around any unused crypto info entries
when adding new ones
2017-02-27 13:55:58 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/atoms.c:
* gst/isomp4/atoms.h:
* gst/isomp4/gstqtmux.c:
qtmux: Update modification times when sending the moov
https://bugzilla.gnome.org/show_bug.cgi?id=779422
2017-03-01 16:11:47 -0800 Michael Smith <[email protected]>
* gst/audioparsers/gstsbcparse.h:
sbcparse: Fix up values for allocation enumeration.
https://bugzilla.gnome.org/show_bug.cgi?id=779389
2017-02-28 13:10:50 +0200 George Kiagiadakis <[email protected]>
* gst/rtpmanager/gstrtprtxreceive.c:
rtprtxreceive: fix potential leak of old, unassociated, association
requests
https://bugzilla.gnome.org/show_bug.cgi?id=722560
2017-02-28 15:47:23 +0200 Sebastian Dröge <[email protected]>
* gst/avi/gstavidemux.c:
avidemux: Don't increment -1 / unset indices
CID 1398545
2017-02-28 15:20:31 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Protect against NULL pointer dereference for streams without
caps
CID 1363332
2017-02-28 12:57:02 +0200 Sebastian Dröge <[email protected]>
* gst/rtp/gstrtph263pay.c:
rtph263pay: Free mac on errors
CID 1212149
2017-02-28 12:45:24 +0200 Sebastian Dröge <[email protected]>
* gst/rtp/gstrtpvorbispay.c:
rtpvorbispay: Add missing break to for loop
2017-02-28 11:02:54 +0100 Edward Hervey <[email protected]>
* tests/check/Makefile.am:
check: Fix splitmux test CFLAGS
Needs to know where the gstapp headers are
2017-02-27 21:02:51 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Fix compilation with gcc 7
qtdemux.c: In function ‘qtdemux_parse_samples’:
qtdemux.c:8450:39: error: ‘*’ in boolean context, suggest ‘&&’
instead [-Werror=int-in-bool-context]
if (stream->samples_per_frame * stream->bytes_per_frame) {
~~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~
2017-02-27 21:01:23 +0200 Sebastian Dröge <[email protected]>
* gst/audioparsers/gstmpegaudioparse.c:
mpegaudioparse: Fix compilation with gcc 7
gstmpegaudioparse.c: In function ‘gst_mpeg_audio_parse_reset’:
gstmpegaudioparse.c:209:3: error: ‘memset’ used with length equal to
number of elements without multiplication by element size
[-Werror=memset-elt-size]
memset (mp3parse->xing_seek_table_inverse, 0, 256);
^~~~~~
gstmpegaudioparse.c: In function
‘gst_mpeg_audio_parse_handle_first_frame’:
gstmpegaudioparse.c:951:7: error: ‘memset’ used with length equal to
number of elements without multiplication by element size
[-Werror=memset-elt-size]
memset (mp3parse->xing_seek_table_inverse, 0, 256);
^~~~~~
2017-02-27 19:31:39 +0200 Sebastian Dröge <[email protected]>
* gst/rtp/gstrtpvorbispay.c:
rtpvorbispay: When getting new headers, replace the old version of
them
This prevents storing an infinite amount of e.g. comment headers if
they
come without a new initialization header in front of them. There can
only be one header of each type.
2017-02-27 19:25:35 +0200 Sebastian Dröge <[email protected]>
* tests/check/Makefile.am:
* tests/check/elements/rtp-payloading.c:
rtp-payloading: Add new test for Vorbis renegotiation
Check if encoding, payloading, depayloading and decoding works if the
stream configuration (and thus the headers) change.
2017-02-27 19:24:07 +0200 Sebastian Dröge <[email protected]>
* gst/rtp/gstrtpvorbispay.c:
vorbispay: Only replace headers when receiving a new config header
If we also replace all headers when receiving any possibly following
comments header, we would throw away the config header before being
able
to make use of it.
2017-02-23 12:11:15 +0200 George Kiagiadakis <[email protected]>
* tests/check/Makefile.am:
* tests/check/elements/splitmux.c:
tests: splitmux: add unit test for content with sparse streams
https://bugzilla.gnome.org/show_bug.cgi?id=761086
2017-02-22 11:23:19 +0200 George Kiagiadakis <[email protected]>
* gst/multifile/gstsplitmuxpartreader.c:
splitmuxpartreader: ignore sparse streams when calculating the end
offset of a part
A sparse stream's ending timestamp can be considerably smaller
than the ending timestamps of the other streams, which can lead
to skipping considerable time from the next part.
https://bugzilla.gnome.org/show_bug.cgi?id=761086
2017-02-22 11:21:06 +0200 George Kiagiadakis <[email protected]>
* gst/multifile/gstsplitmuxpartreader.c:
splitmuxpartreader: identify sparse streams
2017-02-25 21:47:03 -0300 Edgard Lima <[email protected]>
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* gst/audioparsers/gstamrparse.c:
* gst/rtp/gstrtpg726depay.c:
* gst/rtp/gstrtpg726depay.h:
* gst/rtp/gstrtpg726pay.c:
* gst/rtp/gstrtpg726pay.h:
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmadepay.h:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtppcmudepay.h:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtppcmupay.h:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexdepay.h:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtp/gstrtpspeexpay.h:
* sys/v4l2/gstv4l2.c:
* sys/v4l2/gstv4l2bufferpool.c:
* sys/v4l2/gstv4l2bufferpool.h:
* sys/v4l2/gstv4l2colorbalance.c:
* sys/v4l2/gstv4l2colorbalance.h:
* sys/v4l2/gstv4l2object.c:
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2src.c:
* sys/v4l2/gstv4l2src.h:
* sys/v4l2/gstv4l2tuner.c:
* sys/v4l2/gstv4l2tuner.h:
* sys/v4l2/gstv4l2vidorient.c:
* sys/v4l2/gstv4l2vidorient.h:
* sys/v4l2/v4l2_calls.c:
* sys/v4l2/v4l2_calls.h:
Update Edgard Lima's email
https://bugzilla.gnome.org/show_bug.cgi?id=779230
2017-02-08 13:36:00 +0000 Andrew <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.h:
rtpjitterbuffer: Don't always reset PTS to 0 after a gap
In function rtp_jitter_buffer_calculate_pts: If gap in incoming RTP
timestamps is more than (3 * jbuf->clock_rate) we call
rtp_jitter_buffer_reset_skew which resets pts to 0. So components down
the pipeline (playes, mixers) just skip frames/samples until pts
becomes
equal to pts before gap.
In version 1.10.2 and before this checking was bypassed for packets
with
"estimated dts", and gaps were handled correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=778341
2017-02-24 15:59:41 +0200 Sebastian Dröge <[email protected]>
* meson.build:
meson: Update version
2017-02-24 15:37:36 +0200 Sebastian Dröge <[email protected]>
* configure.ac:
Back to development
Download
========
https://download.gnome.org/sources/gst-plugins-good/1.11/gst-plugins-good-1.11.91.tar.xz
(3.31M)
sha256sum: 9e78901a22936b4d5fecfe61d8456499c7b4b0f6d83c11767e2854d3932ac6b3
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