Commit 97e108be authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Release 1.7.2

parent 163a67ab
=== release 1.7.2 ===
2016-02-19 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.7.2
2016-02-19 10:31:05 +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:
po: Update translations
2016-02-18 14:31:28 +0000 Julien Isorce <j.isorce@samsung.com>
* pkgconfig/gstreamer-allocators-uninstalled.pc.in:
* pkgconfig/gstreamer-app-uninstalled.pc.in:
* pkgconfig/gstreamer-audio-uninstalled.pc.in:
* pkgconfig/gstreamer-fft-uninstalled.pc.in:
* pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-riff-uninstalled.pc.in:
* pkgconfig/gstreamer-rtp-uninstalled.pc.in:
* pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
* pkgconfig/gstreamer-sdp-uninstalled.pc.in:
* pkgconfig/gstreamer-tag-uninstalled.pc.in:
* pkgconfig/gstreamer-video-uninstalled.pc.in:
uninstalled.pc: add support for non libtool build systems
Currently the .la path is provided which requires to use libtool as
mentioned in the GStreamer manual section-helloworld-compilerun.html.
It is fine as long as the application is built using libtool.
So currently it is not possible to compile a GStreamer application
within gst-uninstalled with CMake or other build system different
than autotools.
This patch allows to do the following in gst-uninstalled env:
gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
gstreamer-video-1.0)
Previously it required to prepend libtool --mode=link
https://bugzilla.gnome.org/show_bug.cgi?id=720778
2016-01-22 18:26:01 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
* gst/typefind/gsttypefindfunctions.c:
typefind: strengthen check for valid H.263 picture layer
Avoids some false positives leading to miss identification:
* Prevent picture start code emulation for the first 2 bytes read
* Add check for valid "picture coding type" and "PB-frames mode" combination
Additionally, change name on confusingly named TR var to what
it is, the layer's PTYPE.
https://bugzilla.gnome.org/show_bug.cgi?id=693263
2015-11-23 15:06:02 +0900 Vineeth T M <vineeth.tm@samsung.com>
* gst/playback/gstdecodebin2.c:
decodebin: return incomplete topology if decode chains' cap could not be obtained
When getting caps of the decode chain, in get_topology, the caps are being
checked if fixed or not. But get_topology will be called when the decode is
chain is being exposed and hence it will always be fixed. Hence removing the
check for fixed caps. Removing gst_pad_get_current_caps for the chain->pad, as
get_pad_caps will again call the same api.
And get_topology can return NULL value if currently shutting down the
pipeline, which on being passed to create message will result in assertion
error. Check if topology is valid before using it
https://bugzilla.gnome.org/show_bug.cgi?id=755918
2016-02-05 10:10:40 +0100 Havard Graff <havard.graff@gmail.com>
* gst-libs/gst/Makefile.am:
rtp: build audio library before rtp
Because audio-enumtypes.h needs to be available for
gstrtpbaseaudiopayload.c
https://bugzilla.gnome.org/show_bug.cgi?id=761949
2016-02-15 21:28:33 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/playback/gstdecodebin2.c:
decodebin: Fix documentation of the autoplug-query signal
2016-01-26 13:54:46 +0100 Stian Selnes <stian@pexip.com>
* gst-libs/gst/video/gstvideoencoder.c:
* tests/check/libs/videoencoder.c:
videoencoder: Fix leak when pre_push does not return OK
https://bugzilla.gnome.org/show_bug.cgi?id=761951
2016-02-11 19:47:04 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/audioresample/resample.c:
resample: avoid overflows
Avoid overflow in rate calculation. This can cause the resampler to
start on the wrong phase after a rate change.
Avoid overflow in cubic fraction calculation. This can cause noise when
dealing with higher samplerates.
2016-02-11 18:01:40 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/audioresample/resample_sse.h:
resample: fix double interpolation sse code
We were only reading 2 filter taps and we need to read 4 to do cubic
interpolation.
2016-02-10 12:48:15 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-converter.c:
audio-converter: make a copy if we can't write in unpack
If we don't have writable memory, make sure to make a copy of the input
samples into a temporary (writable) buffer, even if we are dealing with
a native intermediate format that we don't need to call the unpack
function for.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=761655
2016-02-05 19:15:16 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* tests/check/Makefile.am:
tests: extend the AM_TESTS_ENVIRONMENT from check.mak
To get the CK_DEFAULT_TIMEOUT defined for all tests.
Also replaces a 120 timeout that was set.
https://bugzilla.gnome.org/show_bug.cgi?id=761472
2016-02-05 18:03:07 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* autogen.sh:
* common:
Automatic update of common submodule
From 86e4663 to b64f03f
2016-01-21 09:43:35 +0100 Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>
* ext/pango/gstbasetextoverlay.c:
* ext/pango/gstbasetextoverlay.h:
textoverlay: Expose rendering dimensions as properties.
In order to detect graphical user input on the
textoverlay, the resulting rendering properties
need to be exposed to applications.
Fixes delayx property declaration.
https://bugzilla.gnome.org/show_bug.cgi?id=761251
2016-01-20 15:37:44 +0100 Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>
* ext/pango/gstbasetextoverlay.c:
textoverlay: Do not limit positioning to video area.
The current position property is limited to X,Y positions
in the range of [0, 1]. This patch allows full control
over the overlay position, including partially outside
of the video area.
https://bugzilla.gnome.org/show_bug.cgi?id=761251
2016-01-28 13:29:39 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/audiorate/gstaudiorate.c:
audiorate: Use gst_audio_format_fill_silence() instead of memset with 0 for generating silence
For unsigned formats, silence is not all bits 0.
2016-01-28 13:21:33 +0100 HoonHee Lee <hoonhee.lee@lge.com>
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/video/gstvideodecoder.c:
audio/videodecoder: Minor cleanup of last commit
https://bugzilla.gnome.org/show_bug.cgi?id=761218
2016-01-28 18:06:44 +0900 HoonHee Lee <hoonhee.lee@lge.com>
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/video/gstvideodecoder.c:
audio/videodecoder: use gst_pad_peer_query_caps to make output caps
gst_pad_get_allowed_caps() will return NULL if the srcpad has no peer.
In that case, use gst_pad_peer_query_caps() with template caps as filter
to have negotiated output caps properly before forwarding GAP event.
https://bugzilla.gnome.org/show_bug.cgi?id=761218
2016-01-26 19:23:04 +0100 Thibault Saunier <tsaunier@gnome.org>
* gst/encoding/gstencodebin.c:
encodebin: Allow streamheader update when profile.allow_dynamic_output == FALSE
Some encoders can update the stream header through time (for example
vp8 might do that) but it does not strictly changes the output format.
2016-01-26 14:09:42 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
* gst-libs/gst/video/video-format.h:
video-format: fix GstVideoFormatInfo documentation warnings
Add missing ':' to tile_ws and tile_hs fields documentation to avoid
bad render of these two fields, mark reserved bytes as private to hide
field and avoid gtkdoc warning and add parameters description to
documented macro to avoid gtkdoc warnings.
https://bugzilla.gnome.org/show_bug.cgi?id=761132
2016-01-26 16:56:57 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-converter.c:
* gst-libs/gst/audio/audio-converter.h:
* win32/common/libgstaudio.def:
audio-converter: add reset function
2016-01-26 16:36:41 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-converter.c:
audio-converter: handle NULL input
Allow NULL as input to mean silence samples.
2016-01-26 17:16:52 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-converter.c:
audio-converter: improve _update_config
Allow NULL config to keep the existing parameters.
Fix the docs.
2016-01-26 17:14:20 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-converter.c:
* gst-libs/gst/audio/audio-converter.h:
audio-converter: audio-converter: make some optimized functions
Make optimized functions for generic and passthrough conversion.
2016-01-26 16:34:35 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-quantize.c:
* gst-libs/gst/audio/audio-quantize.h:
audio-quantize: add _reset function
Add a reset function that clears any history.
2016-01-25 17:40:23 +0000 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
* m4/Makefile.am:
* m4/freetype2.m4:
* tests/examples/Makefile.am:
build: remove nonsensical check for freetype
The examples need Gtk+, nothing uses freetype directly.
2016-01-25 16:22:17 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/elements/libvisual.c:
tests: libvisual: make run faster
Reduce resolution, which shouldn't make any difference
to what's tested here. Makes test finish in less than
half the time it took before (8s vs. 21s).
2016-01-25 18:30:30 +0530 Arun Raghavan <git@arunraghavan.net>
* ext/alsa/gstalsasink.c:
alsa: Trivial doc update
alsasink now does more than just raw audio.
2016-01-21 18:30:40 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/playback/gstdecodebin2.c:
decodebin: Correctly expose pads from elements that have directly exposable pads
analyze_new_pad() can return a new decode chain, which might have a new
GstDecodePad in the end. We should use those two for expose_pad() and not the
original ones that were passed to analyze_new_pad().
This fails when having a demuxer element that has raw pads immediately or
if a decoder with raw caps is after an adaptive demuxer.
https://bugzilla.gnome.org/show_bug.cgi?id=760949
2016-01-21 16:08:46 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-converter.c:
audio-converter: ensure correct alignment of samples
Make sure that the data we allocate for our temporary buffers is
properly aligned.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=760938
2016-01-21 10:45:40 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/video/video-color.c:
* gst-libs/gst/video/video-color.h:
video-color: add Adobe RGB primaries and transfer function
2016-01-20 10:19:34 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/video/video-info.c:
video-info: enfore RGB matrix for RGB formats
In gst_video_info_to_caps(), make sure we end up with an RGB matrix for
RGB formats and warn when the GstVideoInfo colorimetry is wrong.
In gst_video_info_from_caps(), fix the GstVideoInfo with an RGB matrix
for RGB formats and warn about inconsistent caps.
See https://bugzilla.gnome.org/show_bug.cgi?id=759624
2016-01-20 10:02:20 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/video/video-converter.c:
video-converter: ignore matrix for RGB formats
For RGB formats, the matrix in the colorimetry (conversion from YUV to
RGB) is irrelevant and we should ignore it and assume the identity
transform for everything we do.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759624
2016-01-19 23:26:57 +0100 Thibault Saunier <tsaunier@gnome.org>
* gst-libs/gst/video/gstvideoencoder.h:
videoencoder: Deprecate GST_VIDEO_ENCODER_FLOW_DROPPED
It was never actually supported or used
https://bugzilla.gnome.org/show_bug.cgi?id=760666
2016-01-19 23:22:35 +0100 Thibault Saunier <tsaunier@gnome.org>
* gst-libs/gst/video/gstvideoencoder.c:
Revert "videoencoder: Release video frame when ->handle return ERROR or DROPPED"
This reverts commit 63517d0ed348784cce4ab4b295c2c0f1b78baa81.
It was wrong ref counting wise and we decided to deprecated DROPPED
return value
https://bugzilla.gnome.org/show_bug.cgi?id=760666
2016-01-18 11:40:36 +0900 Vineeth TM <vineeth.tm@samsung.com>
* tests/check/elements/audioconvert.c:
tests:audioconvert: Fix integer overflow build error
value of 32768L << 16 and 1L << 31 is 2147483648
but it exceeds the positive range of int which is 2147483647
resulting in integer overflow error. Use G_GINT64_CONSTANT instead of L.
https://bugzilla.gnome.org/show_bug.cgi?id=760769
2016-01-19 12:39:22 +0530 Arun Raghavan <git@arunraghavan.net>
* gst-libs/gst/app/gstappsrc.c:
appsrc: Minor documentation cleanup
2016-01-14 23:14:27 +0000 Tim-Philipp Müller <tim@centricular.com>
* tools/gst-play.c:
tools: gst-play: allow setting of flags in serialized foo+bar format
https://bugzilla.gnome.org/show_bug.cgi?id=751901
2015-07-02 17:58:00 +0200 Hugues Fruchet <hugues.fruchet@st.com>
* tools/gst-play.c:
tools: gst-play: add command line options for verbose output and playbin flags
https://bugzilla.gnome.org/show_bug.cgi?id=751901
2016-01-18 15:51:16 +0200 Sebastian Dröge <sebastian@centricular.com>
* win32/common/libgstapp.def:
win32: Update exports
2015-10-15 10:38:16 -0400 Evan Callaway <evan.callaway@ipconfigure.com>
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
Add WAIT_ON_EOS flag to gstappsink.
If set, an appsink that receives an EOS will wait until all of its buffers have been processed before continuing.
https://bugzilla.gnome.org/show_bug.cgi?id=756187
2016-01-16 10:17:50 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/audio/gstaudioencoder.c:
audioencoder: Add note to the documentation about various settings being reset before set_format()
It's quite unexpected behaviour that various subclass settings are just
reset before set_format(). Unfortunately changing this now has the risk
of breaking existing code but we should reconsider this for 2.0.
2016-01-09 04:35:23 +0100 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
* gst/playback/gststreamsynchronizer.c:
streamsynchronizer: Ignore flushing streams [..]
[..] when resetting group start time. In GES, we are usually connected
to the streamsynchronizer on one audio and one video pad.
When seeking the timeline, both nlecompositions often output their flush_start
before any of them has output its flush_stop.
The current code, when receiving the first flush stop was using the
running time of the start of the second composition, which could
be pretty much anything, and means nothing at that point.
This patch is thread-safe, as STREAM_SYNCHRONIZER_LOCK is taken
both when setting flushing and when checking it.
https://bugzilla.gnome.org/show_bug.cgi?id=750013
2016-01-08 18:53:52 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/playback/gstplaybin2.c:
playbin: Only append non-raw and sysmem pad template caps to the autoplug-query result
Otherwise a decoder supporting GL memory will think that all downstream can
support GL memory because of seeing its own template caps.
https://bugzilla.gnome.org/show_bug.cgi?id=758212
2016-01-08 18:37:16 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/playback/gstplaybin2.c:
Revert "playbin: only add the template caps when the result is empty"
This reverts commit 023af2d3b192f8ebf1bd4fe75a22a4adaedc1e05.
https://bugzilla.gnome.org/show_bug.cgi?id=758212
2016-01-15 13:35:22 +0000 Thibault Saunier <tsaunier@gnome.org>
* gst-libs/gst/video/gstvideoencoder.c:
videoencoder: Release video frame when ->handle return ERROR or DROPPED
https://bugzilla.gnome.org/show_bug.cgi?id=760666
2016-01-15 09:50:29 +0100 Edward Hervey <edward@centricular.com>
* gst/playback/gstplaysink.c:
playsink: Properly mark pending blocked pads
When blocking input pads, we also need to properly set the appropriate
pending flag.
Without this, when switching stream types after initial configuration
(like going from Audio+Video to Audio+Video+Sub) playsink would never
wait for *all* input streams to be blocked (it would just wait for the
new input pad (text in this case) to be blocked).
Since the reconfiguration might introduce unlinking/relinking of elements,
we need to ensure that *ALL* input streams are blocked.
Failure to do so would result in having some input streams pushing data
to inactive elements (returning GST_FLOW_FLUSHING) or unlinked pads
(returning GST_FLOW_NOT_LINKED).
A later optimization could involve only blocking the input pads that
might be involved in reconfiguration. But better be safe than sorry for
now :)
2016-01-06 10:12:43 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* tools/gst-device-monitor.c:
gst-device-monitor: Use g_printerr instead of g_error
g_error is meant to be used for programmer errors (causes an abort),
not for expected runtime errors.
2016-01-13 16:32:25 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: replace gst_caps_can_intersect() with is_subset()
Subset check verifies also that all required fields are present
and is mostly commonly used when checking if an element accepts
a certain caps
2016-01-12 11:31:50 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/playback/gstplaybin2.c:
playbin: use subset check instead of intersect
Elements usually require that all fields on their caps are present
on the fixed caps they receive. Using intersection won't verify it,
resort to using is_subset() checks.
https://bugzilla.gnome.org/show_bug.cgi?id=760477
2016-01-12 15:56:36 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-channel-mixer.c:
audio-channel-mixer: round before truncating
Round the result before truncating for int channel mixing.
2016-01-12 15:27:16 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-converter.c:
audio-converter: Avoid conversion when possible
When the input and output formats are the same and in a possible
intermediate format, avoid unpack and pack.
Never do passthrough channel mixing.
Only do dithering and noise shaping in S32 format
2016-01-12 11:43:20 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-channel-mixer.c:
audio-channel-mixer: add more formats
Add support for float and int16 mixing
Remove in-place processing, this simplifies things as we won't be using it.
Don't do clipping for float audio formats
2016-01-12 11:37:17 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-converter.c:
audio-converter: improve processing loop
Process as many samples as we can from the input and return the number
of processed samples from the chain. This simplifies some code.
Fix the IN_WRITABLE handling, don't overwrite the flags.
2016-01-11 18:24:48 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/playback/gstsubtitleoverlay.c:
subtitleoverlay: replace accept-caps with caps query
Those accept caps are actually checking if downstream supports
some particular caps to check if it need to negotiate a different
format. Checking only the next element with accept-caps is not enough
to guarantee that it is supported.
Using a caps query makes it obtain the supported caps for downstream
as a whole instead of only the next element.
2016-01-08 21:27:16 +0200 Sebastian Dröge <sebastian@centricular.com>
* win32/common/libgstaudio.def:
audio: Update exported symbols list
2016-01-08 15:05:38 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* gst/videorate/gstvideorate.c:
videorate: replace accept-caps with a caps query
accept-caps is only a shallow check, it needs to know
whether downstream as a whole accepts the framerate
2016-01-08 16:08:47 +0000 Tim-Philipp Müller <tim@centricular.com>
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: fix up for GstAudioChannelMix rename as well
2016-01-08 17:34:50 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-converter.c:
* gst-libs/gst/audio/audio-converter.h:
* gst/audioconvert/gstaudioconvert.c:
audio-converter: small API tweaks
Pass flags in _converter_new() so that we can configure ourselves
differently depending on some options.
SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'
2016-01-08 17:28:31 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-converter.c:
* gst-libs/gst/audio/audio-converter.h:
audio-converter: prepare API for rate changes
Use the update function to update the sample rates along with the config
once we implement resampling.
2016-01-08 17:17:44 +0100 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/audio/audio-converter.c:
* gst-libs/gst/audio/audio-converter.h:
* gst/audioconvert/gstaudioconvert.c:
audio-convert: simplify API
Simplify the API, we don't need the consumed and produced output
arguments. The caller needs to use the _get_in_frames/get_out_frames API
to check how much input is needed and how much output will be produced.
2016-01-08 17:50:21 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/audio/gstaudioutilsprivate.h:
* gst-libs/gst/video/gstvideoutilsprivate.h:
audio/video: Use G_GNUC_INTERNAL for internal functions
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:
* gst-libs/gst/audio/audio.h:
* win32/common/libgstaudio.def:
audio: GstAudioChannelMix -> GstAudioChannelMixer
Rename the GstAudioChannelMix object to GstAudioChannelMixer because it
looks better and to avoid a conflict with a library in -bad.
2016-01-07 15:24:25 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/playback/gstplaybin2.c:
playbin: Use the caps query instead of accept-caps to detect if a sink accepts caps
accept-caps is only for one element, caps query is recursive. Fixes playback
with totem and other situations.
https://bugzilla.gnome.org/show_bug.cgi?id=760234
2016-01-06 15:49:59 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
* gst-libs/gst/video/gstvideopool.c:
videopool: store videoinfo after choosing the biggest buffer size
Otherwise, pool could be negotiated with a size which will be different
from the one used in allocation which is the GstVideoInfo.
https://bugzilla.gnome.org/show_bug.cgi?id=760222
2016-01-06 12:14:39 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
* gst/videotestsrc/gstvideotestsrc.c:
videotestsrc: add missing break in set_property switch case
To avoid future issue when adding new properties.
https://bugzilla.gnome.org/show_bug.cgi?id=760204
2016-01-06 01:04:31 +0000 Koop Mast <kwm@FreeBSD.org>
* tests/check/elements/audioconvert.c:
tests: audioconvert: fix test compilation with clang
With clang 3.7.1 on FreeBSD:
elements/audioconvert.c:650:12: error: shifting a negative signed value is
undefined [-Werror,-Wshift-negative-value]
(-32 << 16) + (1 << 15), (-32 << 16) - (1 << 15),
~~~ ^
https://bugzilla.gnome.org/show_bug.cgi?id=760134
2016-01-06 01:06:10 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/libs/audiodecoder.c:
* tests/check/libs/audioencoder.c:
* tests/check/libs/rtp.c:
* tests/check/libs/rtpbasepayload.c:
tests: fix indentation of various unit tests
2016-01-05 22:52:34 +0000 Tim-Philipp Müller <tim@centricular.com>
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
docs: add new audio API
2016-01-03 17:21:18 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst-libs/gst/sdp/gstmikey.h:
* gst-libs/gst/video/video-overlay-composition.h:
docs: remove dummy function declarations with G_INLINE_FUNCTION for gtk-doc
gtk-doc can handle static inline functions just fine these days,
there's no need for this stuff any more.
2016-01-03 10:33:53 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/riff/riff-ids.h:
riff: Add missing closing parenthesis to GST_RIFF_WAVE_FORMAT_ANTEX_ADPCME
Apparently this #define is unused.
2016-01-02 23:29:22 +0100 Stefan Sauer <ensonic@users.sf.net>
* gst-libs/gst/riff/riff-ids.h:
riff-ids: remove trailing whitespace
2016-01-02 23:27:44 +0100 Stefan Sauer <ensonic@users.sf.net>
* gst-libs/gst/riff/riff-ids.h:
riff-ids: fix two swapped ids
For these fourcc ids the name and value is swapped. This was causing a warning
when registering the avi ids.
2015-12-31 20:43:28 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/Makefile.am:
sdp: Also reorder SUBDIRS to try even harder to build the RTP library first
2015-12-31 20:41:38 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/Makefile.am:
sdp: The SDP library depends on the RTP library now and is not independent anymore
Fix up the build dependencies.
2015-10-07 18:50:18 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstmikey.c:
* gst-libs/gst/sdp/gstmikey.h:
* gst-libs/gst/sdp/gstsdpmessage.c:
* gst-libs/gst/sdp/gstsdpmessage.h:
* tests/check/libs/sdp.c:
* win32/common/libgstsdp.def:
sdp: add helper fuctions from/to sdp from/to caps
<gstsdpmessage.h>
GstCaps* gst_sdp_media_get_caps_from_media (const GstSDPMedia *media, gint pt);
GstSDPResult gst_sdp_media_set_media_from_caps (const GstCaps* caps, GstSDPMedia *media);
gchar * gst_sdp_make_keymgmt (const gchar *uri, const gchar *base64);
GstSDPResult gst_sdp_message_attributes_to_caps (GstSDPMessage *msg, GstCaps *caps);
GstSDPResult gst_sdp_media_attributes_to_caps (GstSDPMedia *media, GstCaps *caps);
<gstmikey.h>
GstMIKEYMessage * gst_mikey_message_new_from_caps (GstCaps *caps);
gchar * gst_mikey_message_base64_encode (GstMIKEYMessage* msg);
https://bugzilla.gnome.org/show_bug.cgi?id=745880
2015-12-29 18:14:54 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/audioconvert/gstaudioconvert.c:
audioconvert: Pass pointer arrays instead of singleton pointers to gst_audio_converter_samples()
In this specific case it wouldn't cause problems as we only ever access the
first array element, but let's make explicit what is happening here.
CID 1346530 and 1346529
2015-12-29 17:56:21 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/encoding-profile.c:
encoding-profile: Check for FALSE'ness directly, not by comparing with FALSE
2015-12-29 17:54:44 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/pbutils/encoding-profile.c:
encoding-profile: Don't use preset_name string after free
When we run the loop for another time and do not have a preset name, we would
try to print the preset name of a previous iteration that is already freed.
Also move some other variables into the block where they are actually used
to prevent similar mistakes in the future.
CID 1346536
2015-12-29 14:40:04 +0100 Stefan Sauer <ensonic@users.sf.net>
* tests/check/elements/audioconvert.c:
audioconvert: add a test for gap handling
2015-12-29 14:23:59 +0100 Stefan Sauer <ensonic@users.sf.net>
* gst-libs/gst/audio/audio-converter.c:
* tests/check/elements/audioconvert.c:
audioconvert: fix passthrough operation
We did not take the sample size into account. Rearrange the tests to have more
conversion test and an extra test case for passthrough operations.
Fixes #759890
2015-12-29 11:29:31 +0000 Tim-Philipp Müller <tim@centricular.com>
* tools/gst-device-monitor.c:
tools: gst-device-monitor: print uint properties in both decimal and hex
Some values are easier to read and make sense of in hex.
https://bugzilla.gnome.org//show_bug.cgi?id=759780
2015-11-12 14:01:03 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
* gst-libs/gst/video/video-blend.c:
videoblend: special case 1x1 src dims on increment computation
Fix crash with 1x1 overlay pixmap
https://bugzilla.gnome.org/show_bug.cgi?id=757290
2015-12-28 12:28:26 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/typefind/gsttypefindfunctions.c:
typefindfunctions: Make sure that enough data is available in AAC/ADTS typefinder
We would otherwise read beyond the array bounds and crash every now and then.
This was introduced with 5640ba17c8db80976b7718904e4024dcfe9ee1a0.
https://bugzilla.gnome.org/show_bug.cgi?id=759910
2015-12-27 19:41:43 +0100 Stefan Sauer <ensonic@users.sf.net>
* tests/check/elements/audioconvert.c:
tests: remove commented code from audioconvert test
This is just what we have in gst_check_buffer_data().
2015-12-27 19:25:20 +0100 Stefan Sauer <ensonic@users.sf.net>
* gst-libs/gst/audio/audio-converter.c:
audio-converter: code cleanup
Rename samples to num_samples, since we also have samples in chain, but that is
the data pointer. Always use gzize for num_samples. Make the log output a bit
more homogenous.
2015-12-26 11:34:47 +0000 Tim-Philipp Müller <tim@centricular.com>
* tools/gst-device-monitor.c:
tools: gst-device-monitor: print non-string device properties too
2015-12-26 09:43:56 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/audio/audio-channel-mix.c:
* gst-libs/gst/audio/audio-converter.c:
* gst-libs/gst/audio/audio-quantize.c:
audio: Fix some documentation warnings
Remove/rename function parameters and skip some functions that can't
be used by bindings as they are now.
2015-12-26 09:43:51 +0100 Sebastian Dröge <sebastian@centricular.com>