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
Using g_thread_join() in _finalize() handlers may result in a deadlock
joining the current thread when the last reference is held by a signal
handler.
e.g.:
error 'Resource deadlock avoided' during 'pthread_join (pt->system_thread, NULL)'
The backtrace looks like this:
[...]
g_thread_join ()
gst_gl_window_finalize ()
gst_gl_window_x11_finalize ()
g_object_unref ()
g_value_unset ()
g_signal_emit_valist ()
g_signal_emit ()
gst_gl_window_send_mouse_event ()
gst_gl_window_mouse_event_cb ()
g_main_dispatch ()
[..]
g_main_loop_run ()
gst_gl_window_navigation_thread ()
g_thread_proxy ()
start_thread ()
clone ()
When only linking the element, the upload object will be created from
_transform_caps() but will never be unreffed as the only case is in _stop().
Add an unref if non-NULL to a new finalize handler for this case.
It's possible that the window may have been destroyed when a winsys
event comes in for it.
Fixes an assertion in make -C tests/check generic/states.check
