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>
This is needed for cross-compiling without a build machine compiler
available. The option was added in 0.54, but we only need this in
Cerbero and it doesn't affect older versions so it should be ok.
Will just cause a spurious warning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/477>
Identity was ignoring seek and flush events even when using
a single segment. In the end it means that we couldn't compute
buffers running-time and stream time after seeks.
This commits adds support for flushing seeks only as I have no idea
what to do for non flushing ones.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/450>
In reverse playback, buffers are played back from buffer.stop
(buffer.pts + buffer.duration) to buffer.pts running times which
mean that we need to use the buffer end running time as a buffer
timestsamp, not the buffer pts when using a single segment in reverse
playback.
This is now being tested in
`validate.test.identity.reverse_single_segment`
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/450>
In reverse playback, buffers have to be displayed at buffer.stop running
time, otherwise a same set of buffer can't be displayed in the exact opposite
order to forward playback.
For example, seeking a video stream at 1fps with start=0, stop=5s, rate=1.0
will display the following buffers:
b0.pts = 0s, b0.duration = 1s - at running time = 0s
b1.pts = 1s, b1.duration = 1s - at running time = 1s
b2.pts = 2s, b2.duration = 1s - at running time = 2s
b3.pts = 3s, b3.duration = 1s - at running time = 3s
b4.pts = 4s, b4.duration = 1s - at running time = 4s
<wait at EOS for 1second>
Now, playing that reverse with start=0, stop=5s, rate=1.0 has to display
the following buffers:
b0.pts = 4s, b0.duration = 1s - at running time = 0s
b1.pts = 3s, b1.duration = 1s - at running time = 1s
b2.pts = 2s, b2.duration = 1s - at running time = 2s
b3.pts = 1s, b3.duration = 1s - at running time = 3s
b4.pts = 0s, b4.duration = 1s - at running time = 4s
<wait at EOS for 1second>
With the previous code, it reproduced the following:
b0.pts = 4s, b0.duration = 1s - at running time = 1s
b1.pts = 3s, b1.duration = 1s - at running time = 2s
b2.pts = 2s, b2.duration = 1s - at running time = 3s
b3.pts = 1s, b3.duration = 1s - at running time = 4s
b4.pts = 0s, b4.duration = 1s - at running time = 5s
<NO WAIT AT EOS AND POST EOS RIGHT AWAY>
This is being tested with the `validate.launch_pipeline.sink.reverse_playback_clock_waits.*`
set of tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/450>
In reverse playback, buffers are played back from buffer.stop
(buffer.pts + buffer.duration) to buffer.pts, which means that the
position after the buffer is consumed is buffer.pts, not buffer.pts -
buffer.duration.
Without that change, and when `automatic_eos` feature is on,
we were dropping the last buffers as marking the stream EOS one buffer
too soon.
This is now being tested extensively by GstValidate in the
`validate.test.clock_sync.*` set of tests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/450>
