Multiqueue should only be used to cope with:
* decoupling upstream and dowstream threading (i.e. having separate threads
for elementary streams).
* Ensuring individual queues have enough space to cope with upstream interleave
(distance in stream time between co-located samples). This is to guarantee
that we have enough room in each individual queues to provide new data in
each, without being blocked.
* Limit the queue sizes to that interleave distance (and an extra minimal
buffering size). This is to ensure we don't consume too much memory.
Based on that, multiqueue now continuously calculates the input interleave
(per incoming streaming thread). Based on that, it calculates a target
interleave (currently 1.5 x real_interleave + 250ms padding).
If the target interleave is greater than the current max_size.time, it will
update it accordingly (to allow enough margin to not block).
If the target interleave goes down by more than 50%, we re-adjust it once
we know we have gone past a safe distance (2 x current max_size.time).
This mode can only be used for incoming streams that are guaranteed to be
properly timestamped.
Furthermore, we ignore sparse streams when calculating interleave and maximum
size of queues.
For the simplest of use-cases (single stream), multiqueue acts as a single
queue with a time limit of 250ms.
If there are multiple inputs, but each come from a different streaming thread,
the maximum time limit will also end up being 250ms.
On regular files (more than one input stream from the same upstream streaming
thread), it can reduce the total memory used as much as 10x, ending up with
max_size.time around 500ms.
Due to the adaptive nature, it can also cope with changing interleave (which
can happen commonly on some files at startup/pre-roll time)
This will mean a much lower delay before a subtitles track changes take
effect. Also avoids excessive memory usage in many cases.
This will also consider sparse streams as (individually) never full, so
as to avoid blocking all playback due to one sparse stream.
https://bugzilla.gnome.org/show_bug.cgi?id=600648
* Avoid the computation completely if we know we don't need it (not in
sync time mode)
* Make sure we don't override highest time with GST_CLOCK_TIME_NONE on
unlinked pads
* Ensure the high_time gets properly updated if all pads are not linked
* Fix the comparision in the loop whether the target high time is the same
as the current time
* Split wake_up_next_non_linked method to avoid useless calculation
https://bugzilla.gnome.org/show_bug.cgi?id=757353
When preparing a buffering message, don't report 0% if there
is any bytes left in the queue at all. We still have something
to push, so don't tell the app to start buffering - maybe
we'll get more data before actually running dry.
Sometimes filesink cleanup during stop may fail due to fclose error.
In this case object left partial cleanup with no file opened
but still holding old file descriptor.
It's not possible to change location property in a such state,
so next start will cause old file overwrite if 'append' does not set.
According to man page and POSIX standard about fclose behavior(extract):
------------------------------------------------------------------------
The fclose() function shall cause the stream pointed to by stream
to be flushed and the associated file to be closed.
...
Whether or not the call succeeds, the stream shall be disassociated
from the file and any buffer set by the setbuf() or setvbuf()
function shall be disassociated from the stream.
...
The fclose() function shall perform the equivalent of a close()
on the file descriptor that is associated with the stream
pointed to by stream.
After the call to fclose(), any use of stream results
in undefined behavior.
------------------------------------------------------------------------
So file is in 'closed' state no matter if fclose succeed or not.
And cleanup could be continued.
https://bugzilla.gnome.org/show_bug.cgi?id=757596
The input of queue/queue2 might have DTS set, in which cas we want
to take that into account (instead of the PTS) to calculate position
and queue levels.
https://bugzilla.gnome.org/show_bug.cgi?id=756507
In order to accurately determine the amount (in time) of data
travelling in queues, we should use an increasing value.
If buffers are encoded and potentially reordered, we should be
using their DTS (increasing) and not PTS (reordered)
https://bugzilla.gnome.org/show_bug.cgi?id=756507
Previously this code was just blindly setting the cached flow return
of downstream to GST_FLOW_OK when we get a SEGMENT.
The problem is that this can not be done blindly. If downstream was
not linked, the corresponding sinqlequeue source pad thread might be
waiting for the next ID to be woken up upon.
By blindly setting the cached return value to GST_FLOW_OK, and if that
stream was the only one that was NOT_LINKED, then the next time we
check (from any other thread) to see if we need to wake up a source pad
thread ... we won't even try, because none of the cached flow return
are equal to GST_FLOW_NOT_LINKED.
This would result in that thread never being woken up
https://bugzilla.gnome.org/show_bug.cgi?id=756645
This way one can define new tracing probes without changing the core. We are
using our own quark table, as 1) we only want to initialize them if we're
tracing, 2) we want to share them with the tracers.
Instead of a single invoke() function and a 'mask', register to individual
hooks. This avoids one level of indirection and allows us to remove the
hook enums. The message enms are now renamed to hook enums.
Get the number of cpus and scale process cpu-load accordingly. Switch the
cpuload to be per-mille to get smoother graphs. Add a bit more logging and use
the _OBJECT variant.
Keep tracer base class in tracer and move core support into the utils module.
Add a unstable-api guard to the tracer.h so that external modules would need to
acknowledge the status by setting GST_USE_UNSTABLE_API.
