Usually this information is static for the whole stream, and various
container formats store this information inside the headers for the
whole stream.
Having it inside the caps for these cases simplifies code and makes it
possible to express these requirements more explicitly with the caps.
https://bugzilla.gnome.org/show_bug.cgi?id=771376
All the GstAudioClock method declarations required object of GstClock type
as a first argument, but in fact, required GstAudioClock object (runtime
check in function body). Instead of checking type in run-time, we can
change functions declaration, to accept only GstAudioClock methods. Then,
runtime check is not necessary anymore, since always GstAudioClock object
is passed to a function.
https://bugzilla.gnome.org/show_bug.cgi?id=756628
Also the format must be fixed on the default raw caps. If not
gst_video_info_from_caps() will fail and
gst_video_decoder_negotiate_default_caps() return FALSE.
The test simulates the use case where a gap event is received before
the first buffer causing the decoder to fall back to the default caps.
https://bugzilla.gnome.org/show_bug.cgi?id=773103
Seen on the Jenkins CI:
FAILED: subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o
ccache cc '-Isubprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta' '-fdiagnostics-color=always' '-I../subprojects/gst-plugins-base/gst-libs/gst/audio' '-Isubprojects/gst-plugins-base/gst-libs/gst/audio' '-Isubprojects/gst-plugins-base/.' '-I../subprojects/gst-plugins-base/.' '-Isubprojects/gst-plugins-base/gst-libs' '-I../subprojects/gst-plugins-base/gst-libs' '-Isubprojects/gstreamer/libs' '-I../subprojects/gstreamer/libs' '-Isubprojects/gstreamer/.' '-I../subprojects/gstreamer/.' '-pipe' '-Wall' '-Winvalid-pch' '-DHAVE_CONFIG_H' '-msse4.1' '-fPIC' '-O0' '-g' '-fPIC' '-I/usr/include/glib-2.0' '-I/usr/lib/glib-2.0/include' '-pthread' '-Isubprojects/gstreamer/gst' '-MMD' '-MQ' 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o' '-MF' 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o.d' -o 'subprojects/gst-plugins-base/gst-libs/gst/audio/audio_resampler_sse41@sta/audio-resampler-x86-sse41.c.o' -c ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.c
In file included from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h:24:0,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-private.h:23,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-macros.h:25,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.h:23,
from ../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler-x86-sse41.c:24:
../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h:26:39: fatal error: gst/audio/audio-enumtypes.h: No such file or directory
#include <gst/audio/audio-enumtypes.h>
^
compilation terminated.
Workaround source_root being the root directory of all projects
in the subproject case.
Remove now unneeded getpluginsdir and define c++ tests in the same loop.
Bump meson requirement to 0.35
This makes sure that we only build files that need explicit SIMD support
with the relevant CFLAGS. This allows the rest of the code to be built
without, and specific SSE* code is only called after runtime checks for
CPU features.
https://bugzilla.gnome.org/show_bug.cgi?id=729276
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
They are false positive overflows, because coverity doesn't realize that
hours <= 24, minutes < 60 and seconds < 60 in all functions. Also casting the
number 60 (seconds in minute, minutes in hour) to guint64 for the
calculations, in order to avoid overflowing once we allow more than 24-hour
timecodes.
CIDs #1371459, #1371458
_stdint.h is generated by Autotools and we don't really need it.
stdint.h is now available on all supported platforms.
This really only makes a difference for MSVC, which has it starting from
Visual Studio 2015.
Add GST_FD_MEMORY_FLAG_DONT_CLOSE to avoid closing the fd when the
memory is freed. When you can guarantee the lifetime of the fd is
longer than the memory, this can save a dup() call.
Most of them are overflow related and false positives, but coverity can't know
that these can't overflow without us giving it more information. Add some
assertions for this.
One was an actual issue with flags comparison.
CIDs #1369051, #1369050, #1369049, #1369048, #1369045
WAV is too generic to impose more-or-less arbitrary boundaries on the
sample rate and channel count caps. For example, there are 384 kHz WAV
files. Another example: it is in theory possible that somebody puts DSD
data into a WAV file, which will then have a sample rate of ~2.8 MHz.
For this reason, get rid of the rate and channel caps unless they are
fixed values. Downstream anyway usually knows the limitations better.
https://bugzilla.gnome.org/show_bug.cgi?id=761514
gst_rtp_buffer_add_extension_onebyte_header() and
gst_rtp_buffer_add_extension_twobytes_header() can have a const argument for
the actual extension data.
The _pull_sample() and _pull_preroll() functions block
until a sample is available, EOS happens or the pipeline
is shut down (returning NULL in the last two cases).
This adds _try_pull_sample() and _try_pull_preroll()
functions with a timeout argument to specify the maximum
amount of time to wait for a new sample.
To avoid code duplication, wait forever if the timeout is
GST_CLOCK_TIME_NONE and use that to implement
_pull_sample/_pull_preroll with the original behavior.
Add also corresponding action signals "try-pull-sample"
and "try-pull-preroll".
https://bugzilla.gnome.org/show_bug.cgi?id=768852
Remove unnecessary helper struct for callbacks. The bclass
member of the helper struct was not used, so we can just
remove it and the GET_CLASS() call and simplify the whole
affair by passing the depayloader directly to the callback.
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
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
