The correct behaviour of anything stuck in the ->render() function
between ->unlock() and ->unlock_stop() is to call
gst_base_sink_wait_preroll() and only return an error if this returns an
error, otherwise, it must continue where it left off!
https://bugzilla.gnome.org/show_bug.cgi?id=773912
When running in sync-by-running-time mode, pad groups
that have exactly 1 pad and it's not-linked might never
wake up after computing a high time, as the per-pad-group
high time was only recomputed when a pad in the group
advances.
Wake those up using the global multiqueue high-time across
all other groups instead.
https://bugzilla.gnome.org/show_bug.cgi?id=774322
When subtracting queued data sizes from upstream queries
in queue, queue2, downloadbuffer and typefind, clamp the
result to not go negative, in case upstream returned
a nonsense value that's too small (as could happen if
upstream is estimating, or just broken)
Implement handling in basesink to not unconditionally discard
out-of-segment buffers and expose it as a new property on fakesink
(not unconditionally in all basesink based sinks).
The property defaults to FALSE.
https://bugzilla.gnome.org/show_bug.cgi?id=765734
Otherwise downstream will have an inconsistent set of sticky events at this
point, e.g. when a TAG event is pushed and downstream wants to relate it to
the stream by looking at the current STREAM_START event.
https://bugzilla.gnome.org/show_bug.cgi?id=768526
On the first buffer, it's possible that sink_segment is set but
src_segment has not been set yet. If this is the case, we should not
calculate cur_level.time since sink_segment.position may be large and
src_segment.position default is 0, with the resulting diff being larger
than max-size-time, causing the queue to start leaking (if
leaky=downstream).
One potential consequence of this is that the segment event may be
stored on the srcpad before the caps event is pushed downstream, causing
a g_warning ("Sticky event misordering, got 'segment' before 'caps'").
https://bugzilla.gnome.org/show_bug.cgi?id=773096
low/high-watermark are of type double, and given in range 0.0-1.0. This
makes it possible to set low/high watermarks with greater resolution,
which is useful with large multiqueue max sizes and watermarks like 0.5%.
Also adding a test to check the fill and watermark level behavior.
https://bugzilla.gnome.org/show_bug.cgi?id=770628
To make the code clearer, and to facilitate future improvements, introduce
a distinction between the buffering level and the buffering percentage.
Buffering level: the queue's current fill level. The low/high watermarks
are in this range.
Buffering percentage: percentage relative to the low/high watermarks
(0% = low watermark, 100% = high watermark).
To that end, get_percentage() is renamed to get_buffering_level(). Also,
low/high_percent are renamed to low/high_watermark to avoid confusion.
mq->buffering_percent values are now normalized in the 0..100 range for
buffering messages inside update_buffering(), and not just before sending
the buffering message. Finally the buffering level range is parameterized
by adding a new constant called MAX_BUFFERING_LEVEL.
https://bugzilla.gnome.org/show_bug.cgi?id=770628
When calculating the high_time, cache the group value in each singlequeue.
This fixes the issue by which wake_up_next_non_linked() would use the global
high-time to decide whether to wake-up a waiting thread, instead of the group
one, resulting in those threads constantly spinning.
Tidy up a bit the waiting logic while we're at it.
With this patch, we go from 212% playing a 8 audio / 8 video file down to less
than 10% (most of it being the video decoding).
https://bugzilla.gnome.org/show_bug.cgi?id=770225
low/high-watermark are of type double, and given in range 0.0-1.0. This
makes it possible to set low/high watermarks with greater resolution,
which is useful with large queue2 max sizes and watermarks like 0.5%.
Also adding a test to check the fill and watermark level behavior.
https://bugzilla.gnome.org/show_bug.cgi?id=769449
To make the code clearer, and to facilitate future improvements, introduce
a distinction between the buffering level and the buffering percentage.
Buffering level: the queue's current fill level. The low/high watermarks
are in this range.
Buffering percentage: percentage relative to the low/high watermarks
(0% = low watermark, 100% = high watermark).
To that end, get_buffering_percent() is renamed to get_buffering_level(),
and the code at the end that transforms to the buffering percentage is
factored out into a new convert_to_buffering_percent() function. Also,
the buffering level range is parameterized by adding a new constant called
MAX_BUFFERING_LEVEL.
https://bugzilla.gnome.org/show_bug.cgi?id=769449
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
In ringbuffer mode we need to make sure we post buffering messages *before*
blocking to wait for data to be drained.
Without this, we would end up in situations like this:
* pipeline is pre-rolling
* Downstream demuxer/decoder has pushed data to all sinks, and demuxer thread
is blocking downstream (i.e. not pulling from upstream/queue2).
* Therefore pipeline has pre-rolled ...
* ... but queue2 hasn't filled up yet, therefore the application waits for
the buffering 100% messages before setting the pipeline to PLAYING
* But queue2 can't post that message, since the 100% message will be posted
*after* there is room available for that last buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=769802
Other pads that are waiting for the stream on the selected
pad to advance before they finish waiting themselves
should be given the chance to do so when the selected pad
goes EOS. Fixes problems where input streams can end up
waiting forever if the active stream goes EOS earlier than
their own end time.
When dealing with small-ish input data coming into queue2, such as
adaptivedemux fragments, we would never take into account the last
<200ms of data coming in.
The problem is that usually on TCP connection the download rate
gradually increases (i.e. the rate is lower at the beginning of a
download than it is later on). Combined with small download time (less
than a second) we would end up with a computed average input rate
which was sometimes up to 30-50% off from the *actual* average input
rate for that fragment.
In order to fix this, force the average input rate calculation when
we receive an EOS so that we take into account that final window
of data.
https://bugzilla.gnome.org/show_bug.cgi?id=768649
This is an update on c9b6848885
multiqueue: Fix not-linked pad handling at EOS
While that commit did fix the behaviour if upstream sent a GST_EVENT_EOS,
it would break the same issue when *downstream* returns GST_FLOW_EOS
(which can happen for example when downstream decoders receive data
from after the segment stop).
GST_PAD_IS_EOS() is only TRUE when a GST_EVENT_EOS has flown through it
and not when a GST_EVENT_EOS has gone through it.
In order to handle both cases, also take into account the last flow
return.
https://bugzilla.gnome.org/show_bug.cgi?id=763770
When syncing by running time, multiqueue will throttle unlinked streams
based on a global "high-time" and the pending "next_time" of a stream.
The idea is that we don't want unlinked streams to be "behind" the global
running time of linked streams, so that if/when they get linked (like when
switching tracks) decoding/playback can resume from the same position as
the other streams.
The problem is that it assumes elements downstream will have a more or less
equal buffering/latency ... which isn't the case for streams of different
type. Video decoders tend to have higher latency (and therefore consume more
from upstream to output a given decoded frame) compared to audio ones, resulting
in the computed "high_time" being at the position of the video stream,
much further than the audio streams.
This means the unlinked audio streams end up being quite a bit after the linked
audio streams, resulting in gaps when switching streams.
In order to mitigate this issue, this patch adds a new "group-id" pad property
which allows users to "group" streams together. Calculating the high-time will
now be done not only globally, but also per group. This ensures that within
a given group unlinked streams will be throttled by that group's high-time
instead.
This fixes gaps when switching downstream elements (like switching audio tracks).
Ensure we do not attempt to destroy the current range. Doing so
causes the current one to be left dangling, and it may be dereferenced
later, leading to a crash.
This can happen with a very small queue2 ring buffer (10000 bytes)
and 4 kB buffers.
repro case:
gst-launch-1.0 fakesrc sizetype=2 sizemax=4096 ! \
queue2 ring-buffer-max-size=1000 ! fakesink sync=true
https://bugzilla.gnome.org/show_bug.cgi?id=767688
