These can be passed to gst_type_mark_as_plugin_api, to inform
plugin cache generation.
For now a single flag is specified, "IGNORE_ENUM_MEMBERS", it
can be used for dynamically generated enums to avoid documenting
environment-specific enumeration members. An example is
GstX265EncTune.
Since those are using the clock for sync, they need to also
provide a clock for good measure. The reason is that even if
downstream elements provide a clock, we don't want to have
that clock selected because it might not be running yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/509>
When we want to perform a downstream bitrate query, just
set the reconfigure flag on the srcpad and get the streaming
thread to do it. That avoids emitting a downstream query
when receiving the upstream RECONFIGURE event - which can
lead to deadlocks if downstream is sending the event from
within a lock - e.g. input-selector.
If querying the downstream bitrate changes the cached
value, then make sure to update our buffering state
and potentially post a BUFFERING message to the application.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/566
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/501>
The deadlock was the following:
* One thread requests a new pad, the internal lock is kept while adding the pad
* Another thread (or the same one) requests the internal links of a pad (could
be that pad)... which also requires that lock.
That internal lock is not required when adding the pad to the element (which is
the last action when requesting a new pad). The fact it will be actually used
will be *after* the request pad function is released.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/512>
This can be used to mark additional types exposed by plugins (i.e.
enums, flags and GObjects) via properties, signals or pad templates as
plugin API. They can then be picked up by the documentation for the
plugin.
Not all types exposed by plugins are documented automatically because
they might come from an external library and should be documented from
there instead.
When posting a buffering message succesfully:
* Remember the *actual* percentage value that was posted
* Make sure we only reset the percent_changed variable if the value we just
posted is indeed different from the current value
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/511>
This test takes 39 seconds on my machine even though it just runs
a couple of fakesrc num-buffers=2 ! fakesink pipelines. Most of
the cpu seems to be spent in libz, related to stack trace management.
Use stack-traces-flags=none instead of stack-traces-flags=full
until a better solution can be found. Might warrant more
investigation in any case..
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/503>
This is a follow up to review comments in !297
+ The posting of the buffering message in READY_TO_PAUSED isn't
needed, removing it made the test fail, but the correct fix
was simply to link elements together
+ Move code to relock the queue and set last_posted_buffering_percent
and percent_changed inside the buffering_post_lock in create_write().
This makes locking consistent with post_buffering()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/297>
This fixes a bug that occurs when an attempt is made to post a buffering
message before the queue2 was assigned a bus. One common situation where
this happens is when the use-buffering property is set to TRUE before the
queue2 was added to a bin.
If the result of gst_element_post_message() is not checked, and the
aforementioned situation occurs, then last_posted_buffering_percent and
percent_changed will still be updated, as if posting the message succeeded.
Later attempts to post again will not do anything because the code then
assumes that a message with the same percentage was previously posted
successfully and posting again is redundant.
Updating these variables only if posting succeed and explicitely
posting a buffering message in the READY->PAUSED state change ensure that
a buffering message is posted as early as possible.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/297>
Nowadays we are only waking up the head entry waiting if either the head
entry is unscheduled (which is handled some lines above already), or
when the head entry specifically is woken up because a new entry became
the new head entry.
We're not waking up *all* entries anymore whenever any entry in the last
was unscheduled.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/500>
We already have a mutex in each clock entry anyway and need to make use
of that mutex in most cases when the status changes. Removal of the
atomic operations and usage of the mutex instead simplifies the code
considerably.
The only downside is that unscheduling a clock entry might block for the
time it needs for the waiting thread to go from checking the status of
the entry to actually waiting, which is not a lot of code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/500>
Otherwise it can happen that unscheduling a clock id never takes place
and instead it is waiting until the normal timeout. This can happen if
the wait thread checks the status and sets it to busy, then the
unschedule thread sets it to unscheduled and signals the condition
variable, and then the waiting thread starts waiting. As condition
variables don't have a state (unlike Windows event objects), we have to
remember ourselves in a new boolean flag protected by the entry mutex
whether it is currently signalled, and reset this after waiting.
Previously this was not a problem because a file descriptor was written
to for waking up, and the token was left on the file descriptor until
the read from it for waiting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/500>
gst-tester is a tool to launch `.validatetest` files with
TAP[0] compatible output and supporting missing `gst-validate`
application which means that it can be cleanly integrated with meson
test harness.
It allows us to use `gst-validate` to write integration tests in any
GStreamer repository keeping them as close as possible to the code. It
can simplify a lot test writing and reading and not having to go into
another repository to implement or run tests makes it more convenient to
use.
This also implements a stupid simple test to show how that works
[0] https://testanything.org/
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/461>
This was effectively disabled in 1.0 with the intent of maybe re-enabling it.
The problem is that caching duration at a bin level doesn't make much sense
since there might be queueing/buffering taking place internally and therefore
the duration reported might have no correlation to what is actually being
outputted.
Remove commented code and fixmes, and update documentation
Fixes#4
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/489>
Making it less random and fixing a race in a GES test where we have
as pipeline:
```
videotestsrc ! output-selector name=s ! input-selector name=i s. ! timecodestamper ! i.
```
which we seek, leading to the seek reaching the video testsrc
without going through the timecodestamper and generating a buffer
even before timecodestamper gets the seek which means that its internal
state is wrong compared to the datastream it gets and attaches wrong
timecode metas.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/485>
