...
 
Commits (73)
=== release 1.8.3 ===
2016-08-19 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.8.3
2016-08-19 11:54:35 +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/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:
Update .po files
2016-06-16 10:01:50 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: remove eos avoidance workaround
This workaround tried to avoid an EOS event when seeking to the
end of an Ogg stream in order to find its duration. At some point,
an EOS event there would cause any queue2 upstream to pause and
not restart on a seek back to the beginning. This now appears to
not be the case anymore, and so the workaround can be removed.
https://bugzilla.gnome.org/show_bug.cgi?id=767689
2016-05-24 00:44:21 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst-libs/gst/allocators/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
g-i: pass compiler env to g-ir-scanner
It's what introspection.mak does as well. Should
fix spurious build failures on gnome-continuous.
2016-07-20 11:47:48 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/ogg/gstoggdemux.c:
oggdemux: fix unknown duration playing Ogg over HTTP
If the duration is not known from the chain, it might be known
by the startup seek.
This fixes failure to seek.
Merged with a patch from Tim-Philipp Müller <tim@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=768991
2016-07-18 19:59:23 +1000 Duncan Palmer <dpalmer@digisoft.tv>
* sys/xvimage/xvimageallocator.c:
* sys/xvimage/xvimageallocator.h:
xvimageallocator: const correctness in gst_xvimage_allocator_alloc().
https://bugzilla.gnome.org/show_bug.cgi?id=767712
2016-07-18 14:20:11 +0100 Tim-Philipp Müller <tim@centricular.com>
* sys/xvimage/xvimageallocator.c:
xvimagesink: only error out if the allocated memory is too small
https://bugzilla.gnome.org/show_bug.cgi?id=767712
2016-07-07 22:27:15 +1000 Duncan Palmer <dpalmer@digisoft.tv>
* sys/xvimage/xvimageallocator.c:
* sys/xvimage/xvimageallocator.h:
* sys/xvimage/xvimagepool.c:
xvimagesink: error out on buffer size sanity check failure.
If sanity checks on the buffer size allocated by XvShmCreateImage() fail,
call on g_set_error(), rather than just logging a warning, as this
failure is fatal.
Add a sanity check on buffer size when the video format is RGB. This adds to
existing checks on various YUV pixel formats.
https://bugzilla.gnome.org/show_bug.cgi?id=767712
2016-07-08 16:43:05 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/encoding-profile.c:
encoding-profile: Remove some more fields from the caps when creating from discoverer info
parsed, framed, stream-format and alignment are only relevant for parsers and
should not matter here. We still want to be able to use an encoder that can
only output byte-stream if the input was avc.
https://bugzilla.gnome.org/show_bug.cgi?id=768566
2016-07-08 15:45:25 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/missing-plugins.c:
missing-plugins: Remove some other fields when cleaning up caps
Caps are cleaned up for missing plugins, and for creating encoding profiles
and caps descriptions.
Fields like streamheader, parsed, framed, stream-format and alignment are not
relevant here. The last ones all because a parser will take care of them.
https://bugzilla.gnome.org/show_bug.cgi?id=768566
2016-07-04 17:19:08 +0100 Sergio Torres Soldado <torres.soldado@gmail.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: Fix potential deadlock caused by blocking read forever
Reset the connection "may_cancel" property to avoid a potential deadlock
if there is no data to read and the socket stays blocked forever.
https://bugzilla.gnome.org/show_bug.cgi?id=768249
2016-07-04 11:16:55 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/video/gstvideodecoder.c:
videodecoder: fix criticals fixating a non existent field
https://bugzilla.gnome.org/show_bug.cgi?id=766970
2016-07-04 11:07:54 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: Protect samples_in/bytes_out and audio info with object lock
It might cause invalid calculations during the CONVERT query otherwise.
2016-07-04 11:12:25 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: Protect samples_in/bytes_out and audio info with object lock
It might cause invalid calculations during the CONVERT query otherwise.
2016-07-04 11:00:51 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstaudioutilsprivate.c:
* gst-libs/gst/audio/gstaudioutilsprivate.h:
audioencoder/decoder: Move encoded audio conversion function to a common place
No need to duplicate this non-trivial function.
2016-07-04 09:15:03 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: fix criticals fixating a non existent field
https://bugzilla.gnome.org/show_bug.cgi?id=766970
2016-07-04 10:55:07 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/video/gstvideodecoder.c:
videodecoder: Use the object lock to protect bytes/time tracking
And especially don't use the stream lock for that, as otherwise non-serialized
queries (CONVERT) will cause the stream lock to be taken and easily causes the
application to deadlock.
https://bugzilla.gnome.org/show_bug.cgi?id=768361
2016-07-04 10:47:36 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/video/gstvideodecoder.c:
* gst-libs/gst/video/gstvideoencoder.c:
* gst-libs/gst/video/gstvideoutilsprivate.c:
* gst-libs/gst/video/gstvideoutilsprivate.h:
videoencoder/decoder: Move conversion utility functions to a common header and use consistently in encoder/decoder
2016-07-04 10:52:24 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/video/gstvideoencoder.c:
videoencoder: Use the object lock to protect bytes/time tracking
2016-06-30 18:53:07 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst-libs/gst/tag/gsttagdemux.c:
tagdemux: fix handling of very short files in push mode
By default we'll wait for a certain amount of data before
attempting typefinding. However, if the stream is fairly
short, we might get EOS before we ever attempted any
typefinding, so at this point we should force typefinding
and output any pending data if we manage to detect the
type.
https://bugzilla.gnome.org//show_bug.cgi?id=768178
2016-06-30 17:30:34 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst-libs/gst/tag/gsttagdemux.c:
tagdemux: fix erroring out if we reach EOS without detecting type
In 0.10 the source pad was a dynamic pad that was only added once
the type had been detected, but in 1.x it's an always source pad,
so checking whether it's still NULL won't work to detect if the
type has been detected.
Makes tagdemux error out when we get EOS but haven't managed to
identify the format of the data after the tag.
https://bugzilla.gnome.org//show_bug.cgi?id=768178
2016-06-29 18:14:51 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/audio/audio-channels.c:
* gst/audioconvert/gstaudioconvert.c:
audioconvert: Handle fallback channel mask for mono correctly
It's 0 and no mask should be set for mono at all.
https://bugzilla.gnome.org/show_bug.cgi?id=757472
2016-06-27 20:49:38 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/playback/gstplaysink.c:
playsink: Force STEP events on the video-sink for GST_FORMAT_BUFFERS
It does not make much sense for audio sinks.
2016-06-27 20:53:37 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/playback/gstplaysink.c:
playsink: Don't send another step event to the audio-sink if we got step-done from there
Otherwise we would end up with a deadlock as the audio-sink emits step-done
from its streaming thread.
2016-06-21 10:24:15 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/gstdiscoverer.c:
* tests/check/libs/discoverer.c:
discoverer: Only allow serializing OK discoverer infos to GVariants
They will be incomplete otherwise and we can't generate the full serialized
information, and instead will crash somewhere on the way.
https://bugzilla.gnome.org/show_bug.cgi?id=767859
=== release 1.8.2 ===
2016-06-09 11:50:43 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-opus.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 1.8.2
2016-06-09 11:17:28 +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/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:
Update .po files
2016-06-09 10:05:03 +0300 Sebastian Dröge <sebastian@centricular.com>
* po/hr.po:
* po/pt_BR.po:
* po/sk.po:
po: Update translations
2016-05-10 13:56:13 +0200 Stian Selnes <stian@pexip.com>
* gst-libs/gst/video/video-color.c:
* tests/check/libs/video.c:
video-color: Fix colorimetry IS_UNKNOWN
Fix issue with colorimetry default indicies not being in sync with the
actual table causing IS_UNKNOWN() to sometimes fail.
https://bugzilla.gnome.org/show_bug.cgi?id=767163
2016-05-14 14:41:28 +0200 Olivier Crête <olivier.crete@collabora.com>
* ext/opus/gstopusdec.c:
opusdec: Use GST_AUDIO_DECODER_ERROR
This way, the first invalid stream won't break all decoding.
https://bugzilla.gnome.org/show_bug.cgi?id=766265
2016-05-16 12:52:50 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* gst-libs/gst/video/gstvideosink.c:
videosink: ensure the debug category is always initialized
gst_video_sink_center_rect() can be called without a GstVideoSink
having been instantiated so we can't relly on the video sink
class_init function to init the category.
Fix a warning when running:
GST_CHECKS=test_video_center_rect GST_DEBUG=6 G_DEBUG=fatal_warnings make libs/video.check-norepeat
https://bugzilla.gnome.org/show_bug.cgi?id=766510
2016-05-16 15:39:02 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* gst/playback/gstplaybin2.c:
playbin: fix suburidecodebin leak
We take a ref before removing which was never freeded.
The element is still alive anyway because the group has its own ref as
well.
Fix a leak with the 'test_suburi_error_wrongproto' test.
https://bugzilla.gnome.org/show_bug.cgi?id=766515
2016-05-10 21:34:53 +0900 Hyunjun Ko <zzoon@igalia.com>
* gst-libs/gst/sdp/gstsdpmessage.c:
sdp: parse sdp attributes in case that sdp message doesn't contain mikey message
https://bugzilla.gnome.org/show_bug.cgi?id=766204
2016-04-29 11:06:49 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/encoding-profile.c:
encoding-profile: Fix caps memory leak
2016-04-28 11:18:23 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/encoding-profile.c:
encoding-profile: Fail to create encoding profile from discoverer info if no streams could be added
https://bugzilla.gnome.org/show_bug.cgi?id=765708
2016-04-28 11:21:47 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/encoding-profile.c:
encoding-profile: Recurse into nested container profiles and only add the final audio/video streams
If we e.g. have AVI with DV container with video/audio inside the DV
container, we can't handle this at this point with an encoding profile.
Instead of erroring out, flatten the container hierarchy.
https://bugzilla.gnome.org/show_bug.cgi?id=765708
2016-04-28 11:15:53 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/encoding-profile.c:
encoding-profile: Move adding of each stream to a helper function
https://bugzilla.gnome.org/show_bug.cgi?id=765708
2016-05-02 14:21:55 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/opus/gstopusdec.c:
* tests/check/elements/opus.c:
opusdec: intersect with the filter before returning on getcaps
So upstream gets a smaller set to decide upon as it is what it requested
with the filter
https://bugzilla.gnome.org/show_bug.cgi?id=765684
2016-05-02 10:23:09 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/opus/gstopusdec.c:
* tests/check/elements/opus.c:
opusdec: improve getcaps to return all possible rates
The library is capable of converting to different rates.
Includes tests.
https://bugzilla.gnome.org/show_bug.cgi?id=765684
2016-05-02 10:21:52 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/opus/gstopusdec.c:
opusdec: remove artificial restriction on rate negotiation
Remove restrictions when rate is 48000, the underlying lib supports
converting any of the input to any of the output rates.
https://bugzilla.gnome.org/show_bug.cgi?id=765684
2016-05-01 23:19:57 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* ext/opus/gstopusdec.c:
opusdec: refactor getcaps repeated code into a function
Easier to read and maintain
2016-05-02 10:36:07 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* tests/check/elements/opus.c:
tests: opus: remove apparently useless macro in tests
2016-04-28 09:59:25 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* ext/opus/gstopusdec.c:
opusdec: fix caps leaks
The caps returned by gst_pad_get_allowed_caps() was leaked.
https://bugzilla.gnome.org/show_bug.cgi?id=765706
2016-04-25 17:16:04 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/encoding/gstsmartencoder.c:
smartencoder: Only accept TIME segments for real
... and don't try to push pending data without ever having received a SEGMENT
event before EOS
https://bugzilla.gnome.org/show_bug.cgi?id=765541
2016-04-25 16:47:00 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/codec-utils.c:
codec-utils: H264 level idc 0 is not valid
Don't put level=0 into the caps, it confuses other elements.
https://bugzilla.gnome.org/show_bug.cgi?id=765538
2016-04-25 16:48:36 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/codec-utils.c:
codec-utils: H265 level idc 0 is not valid
Don't put level=0 into the caps, it confuses other elements.
https://bugzilla.gnome.org/show_bug.cgi?id=765538
2016-04-25 16:06:39 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/encoding-profile.c:
encoding-profile: Remove codec_data and streamheader fields from constraint caps
When converting discoverer output to an encoding profile, it makes sense to
omit these. It's very very unlikely that our encoder is going to produce bit
by bit the same codec_data or streamheader.
https://bugzilla.gnome.org/show_bug.cgi?id=765534
2016-04-25 15:05:36 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/encoding-profile.h:
encoding-profile: Don't put G_BEGIN_DECLS around #include statements
It should only be around our own declarations.
=== release 1.8.1 ===
2016-04-20 18:15:39 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-opus.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 1.8.1
2016-04-20 18:02:22 +0300 Sebastian Dröge <sebastian@centricular.com>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.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/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:
Update .po files
2016-04-15 17:48:26 +0100 Tim-Philipp Müller <tim@centricular.com>
* win32/common/libgstsdp.def:
win32: update .def for new API
2016-04-20 15:30:04 +0300 Sebastian Dröge <sebastian@centricular.com>
* po/da.po:
po: Update translations
2016-04-19 17:36:20 +0200 Josep Torra <n770galaxy@gmail.com>
* gst-libs/gst/sdp/gstmikey.c:
* gst-libs/gst/sdp/gstsdpmessage.c:
sdp: update since markers to 1.8.1 for some new APIs
As we decided to backport some fixes we update the since markers.
2016-04-15 00:18:50 -0700 Aleix Conchillo Flaqué <aconchillo@gmail.com>
* gst-libs/gst/sdp/gstsdpmessage.c:
* gst-libs/gst/sdp/gstsdpmessage.h:
sdpmessage: new gst_sdp_media_parse_keymgmt/gst_sdp_media_parse_keymgmt
We add a couple of new functions gst_sdp_media_parse_keymgmt and
gst_sdp_media_parse_keymgmt. We also implement
gst_sdp_message_attributes_to_caps and gst_sdp_media_attributes_to_caps
in terms of these new functions and also gst_mikey_message_to_caps.
2016-04-14 23:29:34 -0700 Aleix Conchillo Flaqué <aconchillo@gmail.com>
* gst-libs/gst/sdp/gstmikey.c:
* gst-libs/gst/sdp/gstmikey.h:
* gst-libs/gst/sdp/gstsdpmessage.c:
mikey: add new function gst_mikey_message_to_caps
2016-04-16 02:11:59 +1000 Jan Schmidt <jan@centricular.com>
* gst-libs/gst/audio/gstaudioringbuffer.c:
Revert "audioringbuffer: start ringbuffer if needed upon commit"
This reverts commit 13ee94ef1091f8a8a90dbd395b39876c26c5188e.
Causes audio glitches at startup by starting to output segments
from the ringbuffer before it has been filled / fully prerolled.
https://bugzilla.gnome.org/show_bug.cgi?id=657076
2016-04-14 17:26:54 -0700 Aleix Conchillo Flaqué <aconchillo@gmail.com>
* gst-libs/gst/sdp/gstmikey.c:
mikey: allow passing srtp or srtcp to create mikey message
Current implementation requires all srtp and srtcp parameters to be
given in the caps. MIKEY uses only one algorithm for encryption and one
for authentication so we now allow passing srtp or srtcp parameters. If
both are given srtp parametres will be preferred.
https://bugzilla.gnome.org/show_bug.cgi?id=765027
2016-04-11 11:28:09 +0200 Fabrice Bellet <fabrice@bellet.info>
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
audio: Fix a race with the audioringbuffer thread
There is a small window of time where the audio ringbuffer thread
can access the parent thread variable, before it's initialized
by the parent thread. The patch replaces this variable use by
g_thread_self().
https://bugzilla.gnome.org/show_bug.cgi?id=764865
2016-03-24 14:59:48 +1100 Jan Schmidt <jan@centricular.com>
* gst/playback/gstdecodebin2.c:
decodebin2: Hold new buffering_post lock while posting msgs
There's a small window between decodebin choosing a buffering level
to post and another thread choosing a different buffering level
where things can race. Close that window by holding a new lock
that's only for posting buffering messages - like what was done
in multiqueue.
https://bugzilla.gnome.org/show_bug.cgi?id=764020
2016-02-26 02:56:15 +1100 Jan Schmidt <jan@centricular.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: Reduce URI typefinder from MAX to LIKELY
Don't claim maximum likelihood for anything that starts
with text that looks like a uri, it's too broad.
=== release 1.8.0 ===
2016-03-24 Sebastian Dröge <slomo@coaxion.net>
2016-03-24 12:19:23 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
releasing 1.8.0
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-opus.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 1.8.0
2016-03-24 11:43:05 +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/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:
Update .po files
2016-03-08 13:22:32 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
......@@ -1234,8 +1961,6 @@
2016-01-08 16:22:25 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio-channel-mix.c:
* gst-libs/gst/audio/audio-channel-mix.h:
* gst-libs/gst/audio/audio-channel-mixer.c:
* gst-libs/gst/audio/audio-channel-mixer.h:
* gst-libs/gst/audio/audio-converter.c:
......@@ -2273,8 +2998,6 @@
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiopack.orc:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.h:
* tests/check/Makefile.am:
* win32/common/libgstaudio.def:
......@@ -2291,8 +3014,6 @@
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioconvert/gstchannelmix.c:
* gst/audioconvert/gstchannelmix.h:
* win32/common/libgstaudio.def:
audio-channel-mix: move channel mixer to audio libs
Move the channel mixer code to the audio library
......@@ -2376,8 +3097,6 @@
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
* gst/audioconvert/gstaudioquantize.h:
* gst/audioconvert/gstfastrandom.h:
audioconvert: move audio quantize code to libs
Move the audio quantize code from audioconvert to the audio library.
......@@ -3248,8 +3967,6 @@
2015-10-01 11:55:59 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
* ext/libvisual/Makefile.am:
* ext/libvisual/gstaudiovisualizer.c:
* ext/libvisual/gstaudiovisualizer.h:
* ext/libvisual/visual.h:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/gstaudiovisualizer.c:
......@@ -11514,8 +12231,6 @@
2014-10-30 11:43:52 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/resampler.c:
* gst-libs/gst/video/resampler.h:
* gst-libs/gst/video/video-converter.c:
* gst-libs/gst/video/video-resampler.c:
* gst-libs/gst/video/video-resampler.h:
......@@ -12093,8 +12808,6 @@
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video-converter.c:
* gst-libs/gst/video/video-converter.h:
* gst-libs/gst/video/video-convertor.c:
* gst-libs/gst/video/video-convertor.h:
* gst-libs/gst/video/video.h:
* gst/videoconvert/gstvideoconvert.c:
* gst/videoconvert/gstvideoconvert.h:
......@@ -12117,7 +12830,6 @@
* gst/videoconvert/gstvideoconvertorc-dist.c:
* gst/videoconvert/gstvideoconvertorc-dist.h:
* gst/videoconvert/gstvideoconvertorc.orc:
* gst/videoconvert/videoconvert.c:
* gst/videoconvert/videoconvert.h:
* tests/check/Makefile.am:
* win32/common/libgstvideo.def:
......@@ -20631,7 +21343,6 @@
* tests/check/Makefile.am:
* tests/check/elements/playbin-complex.c:
* tests/check/elements/playbin-compressed.c:
playbin: Rename compressed unit test to complex
It's not really about compressed streams anymore, but also
about stream switching and stream combiners.
......@@ -23500,7 +24211,6 @@
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/app.h:
* gst-libs/gst/app/gstapp.h:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudio.h:
......@@ -23516,10 +24226,8 @@
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/gstpbutils.h:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/riff/gstriff.h:
* gst-libs/gst/riff/riff.h:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtp.h:
* gst-libs/gst/rtp/rtp.h:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/rtsp.h:
......@@ -23527,7 +24235,6 @@
* gst-libs/gst/sdp/gstsdp.h:
* gst-libs/gst/sdp/sdp.h:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gsttag.h:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/gstvideo.h:
......@@ -27031,7 +27738,6 @@
* tools/.gitignore:
* tools/Makefile.am:
* tools/gst-discoverer-1.0.1:
* tools/gst-discoverer.1.in:
* tools/gst-visualise-m.m:
* tools/gst-visualise.1.in:
tools: remove gst-visualise script
......@@ -28034,9 +28740,6 @@
* gst-libs/gst/video/video-orc-dist.c:
* gst-libs/gst/video/video-orc-dist.h:
* gst-libs/gst/video/video-orc.orc:
* gst-libs/gst/video/videoblendorc-dist.c:
* gst-libs/gst/video/videoblendorc-dist.h:
* gst-libs/gst/video/videoblendorc.orc:
orc: rename to video-orc*
2012-07-23 14:23:39 +0200 Robert Swain <robert.swain@collabora.co.uk>
......@@ -28402,7 +29105,6 @@
2012-07-16 21:58:23 +0200 Stefan Sauer <ensonic@users.sf.net>
* ext/libvisual/Makefile.am:
* ext/libvisual/gstaudiobasevisualizer.c:
* ext/libvisual/gstaudiobasevisualizer.h:
* ext/libvisual/gstaudiovisualizer.c:
* ext/libvisual/gstaudiovisualizer.h:
......@@ -28686,7 +29388,6 @@
* ext/libvisual/Makefile.am:
* ext/libvisual/gstaudiobasevisualizer.c:
* ext/libvisual/gstaudiobasevisualizer.h:
* ext/libvisual/gstbaseaudiovisualizer.c:
* ext/libvisual/gstbaseaudiovisualizer.h:
* ext/libvisual/visual.c:
* ext/libvisual/visual.h:
......@@ -31503,8 +32204,6 @@
* gst-libs/gst/interfaces/.gitignore:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
* gst-libs/gst/interfaces/navigation.c:
* gst-libs/gst/interfaces/navigation.h:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/interfaces/tuner.h:
* gst-libs/gst/interfaces/tunerchannel.c:
......@@ -33448,7 +34147,6 @@
* tests/examples/playback/Makefile.am:
* tests/examples/playback/playback-test.c:
* tests/examples/playback/seek.c:
playback: Rename file from seek.c to playback-test.c
2012-03-02 11:57:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
......@@ -33458,7 +34156,6 @@
* tests/examples/playback/Makefile.am:
* tests/examples/playback/seek.c:
* tests/examples/seek/Makefile.am:
* tests/examples/seek/seek.c:
examples: Move seek example into its own directory
2012-03-02 11:01:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
......@@ -35750,21 +36447,6 @@
* configure.ac:
* docs/plugins/Makefile.am:
* ext/Makefile.am:
* ext/gio/Makefile.am:
* ext/gio/gstgio.c:
* ext/gio/gstgio.h:
* ext/gio/gstgiobasesink.c:
* ext/gio/gstgiobasesink.h:
* ext/gio/gstgiobasesrc.c:
* ext/gio/gstgiobasesrc.h:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiosrc.h:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsink.h:
* ext/gio/gstgiostreamsrc.c:
* ext/gio/gstgiostreamsrc.h:
* gst/gio/Makefile.am:
* gst/gio/gstgio.c:
* gst/gio/gstgio.h:
......@@ -35926,8 +36608,6 @@
* docs/plugins/Makefile.am:
* gst/tcp/Makefile.am:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gstmultisocketsink.c:
* gst/tcp/gstmultisocketsink.h:
* gst/tcp/gsttcp-marshal.list:
......@@ -37902,11 +38582,9 @@
2011-12-20 10:08:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/design/design-audiosinks.txt:
* docs/design/draft-media-types.txt:
* docs/design/part-interlaced-video.txt:
* docs/design/part-mediatype-video-raw.txt:
* docs/design/part-playbin.txt:
* docs/design/part-playbin2.txt:
docs: small update to design docs
2011-12-19 23:41:25 +0100 Stefan Sauer <ensonic@users.sf.net>
......@@ -38897,14 +39575,6 @@
* gst-libs/gst/audio/streamvolume.h:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.c:
* gst-libs/gst/interfaces/mixertrack.h:
* gst-libs/gst/interfaces/streamvolume.c:
* gst-libs/gst/interfaces/streamvolume.h:
* gst/playback/Makefile.am:
* gst/playback/gstplaybin2.c:
* gst/volume/gstvolume.c:
......@@ -38942,14 +39612,6 @@
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/colorbalancechannel.c:
* gst-libs/gst/interfaces/colorbalancechannel.h:
* gst-libs/gst/interfaces/videoorientation.c:
* gst-libs/gst/interfaces/videoorientation.h:
* gst-libs/gst/interfaces/videooverlay.c:
* gst-libs/gst/interfaces/videooverlay.h:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/colorbalance.c:
* gst-libs/gst/video/colorbalance.h:
......@@ -40308,7 +40970,6 @@
* tests/check/Makefile.am:
* tests/check/libs/.gitignore:
* tests/check/libs/audiocdsrc.c:
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/gstlibscpp.cc:
* tests/check/libs/libsabi.c:
* tests/check/libs/struct_arm.h:
......@@ -40337,8 +40998,6 @@
* gst-libs/gst/audio/gstaudiocdsrc.c:
* gst-libs/gst/audio/gstaudiocdsrc.h:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/cdda/gstcddabasesrc.c:
* gst-libs/gst/cdda/gstcddabasesrc.h:
* gst-plugins-base.spec.in:
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-cdda-uninstalled.pc.in:
......@@ -40541,12 +41200,6 @@
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
* gst-libs/gst/rtp/gstrtpbaseaudiopayload.c:
* gst-libs/gst/rtp/gstrtpbaseaudiopayload.h:
* gst-libs/gst/rtp/gstrtpbasedepayload.c:
......@@ -40582,10 +41235,6 @@
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
rename baseaudio* -> audiobase*
2011-11-11 11:52:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
......@@ -40615,8 +41264,6 @@
* gst-libs/gst/audio/gstaudioringbuffer.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/audio/gstringbuffer.h:
rename files to match contained objects
2011-11-11 11:21:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
......@@ -41508,8 +42155,6 @@
* ext/theora/gsttheoradec.c:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/gstmetavideo.c:
* gst-libs/gst/video/gstmetavideo.h:
* gst-libs/gst/video/gstvideometa.c:
* gst-libs/gst/video/gstvideometa.h:
* gst-libs/gst/video/gstvideopool.h:
......@@ -41760,7 +42405,6 @@
* tests/examples/audio/volume.c:
* tests/examples/volume/.gitignore:
* tests/examples/volume/Makefile.am:
* tests/examples/volume/volume.c:
volume: move volume example to audio
2011-10-27 09:42:36 +0200 Stefan Sauer <ensonic@users.sf.net>
......@@ -43342,9 +43986,6 @@
* gst-libs/gst/audio/gstaudiodecoder.h:
* gst-libs/gst/audio/gstaudioencoder.c:
* gst-libs/gst/audio/gstaudioencoder.h:
* gst-libs/gst/audio/gstbaseaudiodecoder.c:
* gst-libs/gst/audio/gstbaseaudiodecoder.h:
* gst-libs/gst/audio/gstbaseaudioencoder.c:
* gst-libs/gst/audio/gstbaseaudioencoder.h:
* win32/common/libgstaudio.def:
audio: rename GstBaseAudioDecoder/Encoder to GstAudioDecoder/Encoder
......@@ -45720,22 +46361,16 @@
* tests/examples/overlay/.gitignore:
* tests/examples/overlay/Makefile.am:
* tests/examples/overlay/gtk-videooverlay.c:
* tests/examples/overlay/gtk-xoverlay.c:
* tests/examples/overlay/qt-videooverlay.cpp:
* tests/examples/overlay/qt-xoverlay.cpp:
* tests/examples/overlay/qtgv-videooverlay.cpp:
* tests/examples/overlay/qtgv-videooverlay.h:
* tests/examples/overlay/qtgv-xoverlay.cpp:
* tests/examples/overlay/qtgv-xoverlay.h:
* tests/examples/seek/jsseek.c:
* tests/examples/seek/seek.c:
* tests/icles/.gitignore:
* tests/icles/Makefile.am:
* tests/icles/stress-videooverlay.c:
* tests/icles/stress-xoverlay.c:
* tests/icles/test-colorkey.c:
* tests/icles/test-videooverlay.c:
* tests/icles/test-xoverlay.c:
tests: update for GstXOverlay => GstVideoOverlay
2011-08-08 10:44:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
......@@ -45753,7 +46388,6 @@
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/videooverlay.c:
* gst-libs/gst/interfaces/videooverlay.h:
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/interfaces/xoverlay.h:
* gst-plugins-base.spec.in:
interfaces: rename GstXOverlay interface to GstVideoOverlay
......@@ -46638,7 +47272,6 @@
2011-07-07 21:24:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* Android.mk:
* android/ffmpegcolorspace.mk:
* android/videoconvert.mk:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* ext/ogg/gstoggmux.c:
......@@ -47460,14 +48093,6 @@
* configure.ac:
* gst/colorspace/Makefile.am:
* gst/colorspace/colorspace.c:
* gst/colorspace/colorspace.h:
* gst/colorspace/colorspace.vcproj:
* gst/colorspace/gstcolorspace.c:
* gst/colorspace/gstcolorspace.h:
* gst/colorspace/gstcolorspaceorc-dist.c:
* gst/colorspace/gstcolorspaceorc-dist.h:
* gst/colorspace/gstcolorspaceorc.orc:
* gst/videoconvert/Makefile.am:
* gst/videoconvert/gstvideoconvert.c:
* gst/videoconvert/gstvideoconvert.h:
......@@ -48482,7 +49107,6 @@
* tests/check/elements/decodebin2.c:
* tests/check/elements/playbin-compressed.c:
* tests/check/elements/playbin.c:
* tests/check/elements/playbin2-compressed.c:
* tests/check/elements/playbin2.c:
tests: fix up unit tests for playbin2/decodebin2 renames and updates
Even if they don't work yet.
......@@ -49264,7 +49888,6 @@
* configure.ac:
* gst-libs/gst/audio/.gitignore:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/testchannels.c:
* tests/examples/Makefile.am:
* tests/examples/audio/.gitignore:
* tests/examples/audio/Makefile.am:
......@@ -55133,7 +55756,6 @@
2010-12-13 09:58:53 +0200 Stefan Kost <ensonic@users.sf.net>
* docs/design-audiosinks.txt:
* docs/design/design-audiosinks.txt:
docs: move design doc to design folder
......@@ -56421,7 +57043,6 @@
* docs/libs/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/descriptions.c:
* gst-libs/gst/pbutils/gstdiscoverer-private.h:
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/missing-plugins.c:
......@@ -56834,7 +57455,6 @@
* gst-libs/gst/pbutils/pbutils.h:
* gst/typefind/Makefile.am:
* gst/typefind/gstaacutil.c:
* gst/typefind/gstaacutil.h:
* gst/typefind/gsttypefindfunctions.c:
* win32/common/libgstpbutils.def:
pbutils: add codec-specific utility functions for AAC
......@@ -58241,16 +58861,6 @@
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/playback/.gitignore:
* tests/examples/playback/Makefile.am:
* tests/examples/playback/decodetest.c:
* tests/examples/playback/test.c:
* tests/examples/playback/test2.c:
* tests/examples/playback/test3.c:
* tests/examples/playback/test4.c:
* tests/examples/playback/test5.c:
* tests/examples/playback/test6.c:
* tests/examples/playback/test7.c:
* tests/icles/Makefile.am:
* tests/icles/playback/.gitignore:
* tests/icles/playback/Makefile.am:
......@@ -58318,14 +58928,6 @@
* configure.ac:
* gst/playback/.gitignore:
* gst/playback/Makefile.am:
* gst/playback/decodetest.c:
* gst/playback/test.c:
* gst/playback/test2.c:
* gst/playback/test3.c:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/playback/test7.c:
* tests/examples/Makefile.am:
* tests/examples/playback/.gitignore:
* tests/examples/playback/Makefile.am:
......@@ -62287,7 +62889,6 @@
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/tag/lang.c:
* gst/ffmpegcolorspace/Makefile.am:
* gst/ffmpegcolorspace/gstffmpeg.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
......@@ -63964,10 +64565,6 @@
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoraenc.c:
* ext/theora/gsttheoraparse.c:
* ext/theora/theora.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c:
theora: Rename source files to have the same name as the headers
2010-01-14 10:07:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
......@@ -63978,11 +64575,6 @@
* ext/vorbis/gstvorbisenc.c:
* ext/vorbis/gstvorbisparse.c:
* ext/vorbis/gstvorbistag.c:
* ext/vorbis/vorbis.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
vorbis: Rename source files to have the same name as the headers
2010-01-14 10:05:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
......@@ -64097,7 +64689,6 @@
2010-01-07 15:26:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/lang-tables.c:
* gst-libs/gst/tag/lang-tables.dat:
* gst-libs/gst/tag/lang.c:
tag: fix up disting of lang-tables.c more correctly
......@@ -65821,7 +66412,6 @@
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtsp-marshal.list:
* gst-libs/gst/rtsp/gstrtspextension.c:
* gst-libs/gst/rtsp/rtsp-marshal.list:
* gst-libs/gst/video/Makefile.am:
* gst/playback/Makefile.am:
* gst/tcp/Makefile.am:
......@@ -72832,13 +73422,9 @@
* ext/vorbis/gstvorbistag.h:
* ext/vorbis/vorbis.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisdec.h:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisenc.h:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbisparse.h:
* ext/vorbis/vorbistag.c:
* ext/vorbis/vorbistag.h:
vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
......@@ -73547,22 +74133,9 @@
* gst/audioresample/speex_resampler_int.c:
* gst/audioresample/speex_resampler_wrapper.h:
* gst/speexresample/Makefile.am:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/fixed_arm4.h:
* gst/speexresample/fixed_arm5e.h:
* gst/speexresample/fixed_bfin.h:
* gst/speexresample/fixed_debug.h:
* gst/speexresample/fixed_generic.h:
* gst/speexresample/gstspeexresample.c:
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/resample_sse.h:
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
* gst/typefind/gsttypefindfunctions.c:
* tests/check/Makefile.am:
* tests/check/elements/audioresample.c:
# GStreamer 1.8 Release Notes
**GStreamer 1.8.0 was released on 24 March 2016.**
The GStreamer team is proud to announce a new major feature release in the
stable 1.x API series of your favourite cross-platform multimedia framework!
As always, this release is again packed with new features, bug fixes and other
improvements.
See [https://gstreamer.freedesktop.org/releases/1.8/][latest] for the latest
version of this document.
*Last updated: Thursday 24 March 2016, 10:00 UTC [(log)][gitlog]*
[latest]: https://gstreamer.freedesktop.org/releases/1.8/
[gitlog]: https://cgit.freedesktop.org/gstreamer/www/log/src/htdocs/releases/1.8/release-notes-1.8.md
## Highlights
- **Hardware-accelerated zero-copy video decoding on Android**
- **New video capture source for Android using the android.hardware.Camera API**
- **Windows Media reverse playback** support (ASF/WMV/WMA)
- **New tracing system** provides support for more sophisticated debugging tools
- **New high-level GstPlayer playback convenience API**
- **Initial support for the new [Vulkan][vulkan] API**, see
[Matthew Waters' blog post][vulkan-in-gstreamer] for more details
- **Improved Opus audio codec support**: Support for more than two channels; MPEG-TS demuxer/muxer can now handle Opus;
[sample-accurate][opus-sample-accurate] encoding/decoding/transmuxing with
Ogg, Matroska, ISOBMFF (Quicktime/MP4), and MPEG-TS as container;
[new codec utility functions for Opus header and caps handling][opus-codec-utils]
in pbutils library. The Opus encoder/decoder elements were also moved to
gst-plugins-base (from -bad), and the opus RTP depayloader/payloader to -good.
[opus-sample-accurate]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiometa.html#GstAudioClippingMeta
[opus-codec-utils]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstpbutilscodecutils.html
- **GStreamer VAAPI module now released and maintained as part of the GStreamer project**
[vulkan]: https://www.khronos.org/vulkan
[vulkan-in-gstreamer]: http://ystreet00.blogspot.co.uk/2016/02/vulkan-in-gstreamer.html
## Major new features and changes
### Noteworthy new API, features and other changes
- New GstVideoAffineTransformationMeta meta for adding a simple 4x4 affine
transformation matrix to video buffers
- [g\_autoptr()](https://developer.gnome.org/glib/stable/glib-Miscellaneous-Macros.html#g-autoptr)
support for all types is exposed in GStreamer headers now, in combination
with a sufficiently-new GLib version (i.e. 2.44 or later). This is primarily
for the benefit of application developers who would like to make use of
this, the GStreamer codebase itself will not be using g_autoptr() for
the time being due to portability issues.
- GstContexts are now automatically propagated to elements added to a bin
or pipeline, and elements now maintain a list of contexts set on them.
The list of contexts set on an element can now be queried using the new functions
[gst\_element\_get\_context()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-get-context)
and [gst\_element\_get\_contexts()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-get-contexts). GstContexts are used to share context-specific configuration objects
between elements and can also be used by applications to set context-specific
configuration objects on elements, e.g. for OpenGL or Hardware-accelerated
video decoding.
- New [GST\_BUFFER\_DTS\_OR\_PTS()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html#GST-BUFFER-DTS-OR-PTS:CAPS)
convenience macro that returns the decode timestamp if one is set and
otherwise returns the presentation timestamp
- New GstPadEventFullFunc that returns a GstFlowReturn instead of a gboolean.
This new API is mostly for internal use and was added to fix a race condition
where occasionally internal flow error messages were posted on the bus when
sticky events were propagated at just the wrong moment whilst the pipeline
was shutting down. This happened primarily when the pipeline was shut down
immediately after starting it up. GStreamer would not know that the reason
the events could not be propagated was because the pipeline was shutting down
and not some other problem, and now the flow error allows GStreamer to know
the reason for the failure (and that there's no reason to post an error
message). This is particularly useful for queue-like elements which may need
to asynchronously propagate a previous flow return from downstream.
- Pipeline dumps in form of "dot files" now also show pad properties that
differ from their default value, the same as it does for elements. This is
useful for elements with pad subclasses that provide additional properties,
e.g. videomixer or compositor.
- Pad probes are now guaranteed to be called in the order they were added
(before they were called in reverse order, but no particular order was
documented or guaranteed)
- Plugins can now have dependencies on device nodes (not just regular files)
and also have a prefix filter. This is useful for plugins that expose
features (elements) based on available devices, such as the video4linux
plugin does with video decoders on certain embedded systems.
- gst\_segment\_to\_position() has been deprecated and been replaced by the
better-named gst\_segment\_position\_from\_running\_time(). At the same time
gst\_segment\_position\_from\_stream\_time() was added, as well as \_full()
variants of both to deal with negative stream time.
- GstController: the interpolation control source gained a new monotonic cubic
interpolation mode that, unlike the existing cubic mode, will never overshoot
the min/max y values set.
- GstNetAddressMeta: can now be read from buffers in language bindings as well,
via the new gst\_buffer\_get\_net\_address\_meta() function
- ID3 tag PRIV frames are now extraced into a new GST\_TAG\_PRIVATE\_DATA tag
- gst-launch-1.0 and gst\_parse\_launch() now warn in the most common case if
a dynamic pad link could not be resolved, instead of just silently
waiting to see if a suitable pad appears later, which is often perceived
by users as hanging -- they are now notified when this happens and can check
their pipeline.
- GstRTSPConnection now also parses custom RTSP message headers and retains
them for the application instead of just ignoring them
- rtspsrc handling of authentication over tunneled connections (e.g. RTSP over HTTP)
was fixed
- gst\_video\_convert\_sample() now crops if there is a crop meta on the input buffer
- The debugging system printf functions are now exposed for general use, which
supports special printf format specifiers such as GST\_PTR\_FORMAT and
GST\_SEGMENT\_FORMAT to print GStreamer-related objects. This is handy for
systems that want to prepare some debug log information to be output at a
later point in time. The GStreamer-OpenGL subsystem is making use of these
new functions, which are [gst\_info\_vasprintf()][gst_info_vasprintf],
[gst\_info\_strdup\_vprintf()][gst_info_strdup_vprintf] and
[gst\_info\_strdup\_printf()][gst_info_strdup_printf].
- videoparse: "strides", "offsets" and "framesize" properties have been added to
allow parsing raw data with strides and padding that do not match GStreamer
defaults.
- GstPreset reads presets from the directories given in GST\_PRESET\_PATH now.
Presets are read from there after presets in the system path, but before
application and user paths.
[gst_info_vasprintf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-vasprintf
[gst_info_strdup_vprintf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-strdup-vprintf
[gst_info_strdup_printf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-strdup-printf
### New Elements
- [netsim](): a new (resurrected) element to simulate network jitter and
packet dropping / duplication.
- New VP9 RTP payloader/depayloader elements: rtpvp9pay/rtpvp9depay
- New [videoframe_audiolevel]() element, a video frame synchronized audio level element
- New spandsp-based tone generator source
- New NVIDIA NVENC-based H.264 encoder for GPU-accelerated video encoding on
suitable NVIDIA hardware
- [rtspclientsink](), a new RTSP RECORD sink element, was added to gst-rtsp-server
- [alsamidisrc](), a new ALSA MIDI sequencer source element
### Noteworthy element features and additions
- *identity*: new ["drop-buffer-flags"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-identity.html#GstIdentity--drop-buffer-flags)
property to drop buffers based on buffer flags. This can be used to drop all
non-keyframe buffers, for example.
- *multiqueue*: various fixes and improvements, in particular special handling
for sparse streams such as substitle streams, to make sure we don't overread
them any more. For sparse streams it can be normal that there's no buffer for
a long period of time, so having no buffer queued is perfectly normal. Before
we would often unnecessarily try to fill the subtitle stream queue, which
could lead to much more data being queued in multiqueue than necessary.
- *multiqueue*/*queue*: When dealing with time limits, these elements now use the
new ["GST_BUFFER_DTS_OR_PTS"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html#GST-BUFFER-DTS-OR-PTS:CAPS)
and ["gst_segment_to_running_time_full()"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstSegment.html#gst-segment-to-running-time-full)
API, resulting in more accurate levels, especially when dealing with non-raw
streams (where reordering happens, and we want to use the increasing DTS as
opposed to the non-continuously increasing PTS) and out-of-segment input/output.
Previously all encoded buffers before the segment start, which can happen when
doing ACCURATE seeks, were not taken into account in the queue level calculation.
- *multiqueue*: New ["use-interleave"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-multiqueue.html#GstMultiQueue--use-interleave)
property which allows the size of the queues to be optimized based on the input