It was always wrong since the symbols being exported in gstgl-1.0.dll
are platform-specific, and the check we do in dist checks it on all
platforms (which usually means Linux) and the list is instead
Linux-specific right now.
Even if we fix that, it can still never be right because it depends on
your configuration even on a specific platform. For instance, when we
start building EGL support on Windows using ANGLE, the symbol list will
change depending on whether that is enabled or not.
We also don't need this anymore since we use GST_EXPORT for all
functions exported on Windows now.
MSVC warns about this because it's a C++ compiler, and this actually
results in useful things such as the incorrect 'gboolean' return value
for functions that return GstFlowReturn, so let's do explicit
conversions to reduce the noise and increase its efficacy.
With MSVC, this gives the following warning:
warning C4305: 'function': truncation from 'double' to 'gfloat'
Apparently, MSVC does not figure out what type to use for constants
based on the assignment. This warning is very spammy, so let's try to
fix it.
The error is:
unary minus operator applied to unsigned type, result still unsigned
This is a commonly-done operation in gstreamer and it's done on purpose.
It's just noise.
The headers we include already define boolean on Windows with MSVC, and
it leads to a typedef redefinition error with jpeglib.h which tries to
redefine it in jmorecfg.h
At minimum, we only need to glFlush() if we are in a shared GL context
environment. Move the glFinish() to when the actual wait is requested
which may be never. Improves the throughput on older GL systems without
GL3/GLES3 and/or fence sync objects.
In file included from ../subprojects/gst-plugins-base/gst-libs/gst/video/video.h:27:0,
from ../subprojects/gst-plugins-bad/gst/segmentclip/gstvideosegmentclip.c:25:
../subprojects/gst-plugins-base/gst-libs/gst/video/video-format.h:27:39: fatal error: gst/video/video-enumtypes.h: No such file or directory
#include <gst/video/video-enumtypes.h>
^
compilation terminated.
https://ci.gstreamer.net/job/GStreamer-master-meson/269/console
In M2TS mode, we need an extra 4 bytes in the buffer, so need
to ensure the buffer can contain these. The allocation site
does not know the mode, so this is done in all cases.
* Rephrase tune error to be delsys-neutral
* Refer to the actual check in the 'missing sanity check' warnings
* Use "Delivery system" instead of 'delsys'. The
latter is OK as a shorthand in the code but not
even a real word
This was a regression.
We only have a upstream-id via STREAM_START if we were in push-mode.
In pull-mode we need to create one.
Note: It would be good to eventually have that method (copied from
gst_pad_get_stream_id_internal()) public in the future
For each MpegTSBaseStream, we have a GstStream object which
subclasses can extend with information.
For each program a GstStreamCollection is created with all
GstStream from each stream.
When dealing with TIME-based input, the incoming stream could have
potentially changed completely.
In order to check whether it did or not, we need to re-check all sections
(PAT, PMT...). If it didn't, we will keep using the existing streams/pad,
and if it did we will act as if there was a program switch.
Fixes HLS streaming with decodebin3/playbin3
In order to calculate the *actual* bitrate for downloading a fragment
we need to take into account the time since we requested the fragment.
Without this, the bitrate calculations (previously reported by queue2)
would be biased since they wouldn't take into account the request latency
(that is the time between the moment we request a specific URI and the
moment we receive the first byte of that request).
Such examples were it would be biased would be high-bandwith but high-latency
networks. If you download 5MB in 500ms, but it takes 200ms to get the first
byte, queue2 would report 80Mbit/s (5Mb in 500ms) , but taking the request
into account it is only 57Mbit/s (5Mb in 700ms).
While this would not cause too much issues if the above fragment represented
a much longer duration (5s of content), it would cause issues with short
ones (say 1s, or when doing keyframe-only requests which are even shorter)
where the code would expect to be able to download up to 80Mbit/s ... whereas
if we take the request time into account it's much lower (and we would
therefore end up doing late requests).
Also calculate the request latency for debugging purposes and further
usage (it could allow us to figure out the maximum request rate for
example).
https://bugzilla.gnome.org/show_bug.cgi?id=733959https://bugzilla.gnome.org/show_bug.cgi?id=772330
