Fields related to stream handling (input_streams,
output_streams, slots, guint slot_id) where used totally unprotected
until know.
This lead to several races, especially playing back RTSP streams.
To protect those fields, the OBJECT_LOCK can not be used as we sometimes
need to be able to post message on the bus while holding it.
decodebin3 already has a lock to manage stream selection, and in the end
it makes sense to protect all the stream management fields with the same
lock which is why we reuse the SELECTION_LOCK here.
https://bugzilla.gnome.org/show_bug.cgi?id=784012
decodebin3 checks input streams and pushes EOS if all input streams
are EOSed. If not, fake EOS is pushed to the corresponding slot.
When adaptivedemux is used with multi-track configuration,
adaptivedemux never ever push EOS to non-selected track
because streaming thread for the slot stops with not-linked flow return.
So, decodebin3 should generate EOS itself to finish playback.
https://bugzilla.gnome.org/show_bug.cgi?id=777735
linked input of slot can be old input, so urisourcebin should check
eos state to figure out whether it's new one or not.
If not, urisourcebin never ever forwards EOS to downstream at the end
of presentation, because the old input is still there without removal
https://bugzilla.gnome.org/show_bug.cgi?id=777735
group-id in stream-start event might be updated in
parse_chain_output_probe (). This cause duplicated stream-start
twice with identical stream-id and seq-num, but only group-id is
different. Although there is no change, stream-start event will
be followed by the first buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=771088
This makes it possible for GstDiscoverer to work with sources that
have multiple source pads and hence will trigger the creation of multiple
decodebin instances such as rtspsrc.
Based on the work of Vineeth TM <vineeth.tm@samsung.com>
https://bugzilla.gnome.org/show_bug.cgi?id=754178
The base class is trying to align the processed data, but it endup
removing the GstVideoMeta. That caused wrong result. Instead, just copy
from the process function with the appropriate alignment.
https://bugzilla.gnome.org/show_bug.cgi?id=781204
And only set low-percent/high-percent if not using downloadbuffer, just
like in old uridecodebin. using the watermark based buffering causes
playback to hang never finish buffering with downloadbuffer.
With both audiorate and videorate, it seems more sensible to apply rate
adjustments after the first buffer appears. For example, with v4l2src,
there is often a small delay before the first video buffer turns up, and
this can cause a stuttery start because of videorate trying to ensure a
perfect stream.
Those multiqueue are the ones dealing with adaptive demuxers. They should
have a time limit set so that they don't end up buffering too much data.
They would previously be set with no limits at all, which would cause them
to grow indefinitely until downstream blocks.
gst_video_rate_flush_prev() ensures that the pushed buffer is writable
by calling gst_buffer_make_writable() on videorate->prevbuf.
In drop-only mode we always push buffers directly when they are received
from GstBaseTransform (gst_video_rate_transform_ip()) and do not keep them
around. GstBaseTransform already ensures that those buffers are
writable so there is no need to do it twice.
This change saves us from copying buffers in drop-only mode as we no longer
calls gst_buffer_make_writable() with a buffer having a refcount of 2
(one ref owned by GstBaseTransform and one in videorate->prevbuf).
https://bugzilla.gnome.org/show_bug.cgi?id=780767
Instead go backwards before segment.stop based on the framerate or the
next buffers end timestamp. Otherwise the first buffer will usually be
dropped because outside the segment.
https://bugzilla.gnome.org/show_bug.cgi?id=781899
HLS files can have arbitrary extra tags in them, and
those can be quite long lines. We need to search
further than 256 bytes sometimes just to get past the
first few lines of the file. Make the limit 4KB,
which matches a typical input block size and should
hopefully cover every crazy input.
https://bugzilla.gnome.org/show_bug.cgi?id=780559
The term stride is confusing here, since the stride is always use
to signal the pixel row size of an image (including padding). Also
a frame may have a single stride, which adds to the confusion. This
patch uses frame-size, which simply indicate the frame size in the
case the images have some padding in between.
https://bugzilla.gnome.org/show_bug.cgi?id=780053
This allow using those property through gst-launch-1.0. This type
gained a deserilizer recently. The syntax is: <val1, val2, ...>.
Note that we also use the type int instead of uint to avoid having
to cast when specifying the values. The deserilizers assume
int by default.
https://bugzilla.gnome.org/show_bug.cgi?id=780053
When a clip has video audio and subtitle, if need send gap event
to audio and subtitle, we should make sure all has been sent, so
need every stream keep one send_gap_event.
https://bugzilla.gnome.org/show_bug.cgi?id=780429
When posting 100% buffering due to removing the last
buffering element, we still need to hold the posting
lock as well, to avoid any race with other elements
that might post a buffering message at that exact
moment
Add locking, and handle EOS properly now that urisourcebin
uses custom events in place of real EOS events, so we
need to manually remove buffering messages and potentially
post 100% in that situation
The expanded 4 second buffering was making radio streams that are
being delivered at real-time speeds too slow. We might need
a better plan for matching the queue2 size to incoming bitrate
in the absence of tag information or timestamping.
In uridecodebin, it used tags on the output of decodebin to
adjust the queue2 buffering, but urisourcebin doesn't have that
view - decodebin is downstream from us.
This adds a property to select the maximum number of threads to use for
conversion and scaling. During processing, each plane is split into
an equal number of consecutive lines that are then processed by each
thread.
During tests, this gave up to 1.8x speedup with 2 threads and up to 3.2x
speedup with 4 threads when converting e.g. 1080p to 4k in v210.
https://bugzilla.gnome.org/show_bug.cgi?id=778974
See https://bugzilla.gnome.org/show_bug.cgi?id=773666
This would ideally be solved in baseparse but that requires further
thought at this point, and in the meantime it would be good to have
rawbaseparse not assert on this but handle it gracefully instead.
Probe for MultiQueue source pad might receive EOS twice,
the first is fake-eos and the other is actual EOS.
And the slot can be freed with fake-eos/EOS if the slot has no input.
Since slot freeing is async, double free can be possible.
So, decodebin3 needs to remove the probe also with slot freeing.
https://bugzilla.gnome.org/show_bug.cgi?id=777530
"requested_selection" list might be generated by select-streams event.
And memory of stream-id(s) in select-streams is independent from that of stream-collection.
https://bugzilla.gnome.org/show_bug.cgi?id=775553
The latency query originally had a fallthrough to the default
label at the end as fallback, but that got messed up when the
DURATION and POSITION queries were added, so it then fell through
to the duration query handler instead. Restore original behaviour.
https://bugzilla.gnome.org/show_bug.cgi?id=699077
Duration query would return TRUE and duration=-1. This
worked in the unit test because the unit test implementation
was a bit broken.
Both queries need to access rate with a lock.
Fix broken duration query test as well. It relied on broken
behaviour by the videorate query handler, and also it was
implemented as a downstream query rather than an upstream
query. And we must return HANDLED from the probe so that the
query we intercept actually returns TRUE.
https://bugzilla.gnome.org/show_bug.cgi?id=699077
When the decodebin state change fails because of an error
message, we might not go through PAUSED->READY. Don't leak
a ref to decodebin pads due to pad blocking in that case.
This is because we return ASYNC going to PAUSED, and if
we fail before reaching PAUSED the only transition we'll
see is READY->NULL.
https://bugzilla.gnome.org/show_bug.cgi?id=775893
This adds some extra options that affect pattern=ball mode, allowing the
animation to be synced to running time or wall-time clock for comparing
sync across different instances / pipelines / machines.
Also added is the ability to invert the rendering colours every second,
and some different ball motion patterns.
https://bugzilla.gnome.org/show_bug.cgi?id=740557
The state of urisourcebin (and all elements contained within) can
change at any point in time, including when setting up the typefind
element.
In order to avoid ending up with typefind starting without being fully
connected, lock the state and connect to the 'have-type' signal.
Due to the special nature of adaptivedemux, reconfigure happens
frequently with seek/track-change.
In very exceptional cases, the following sequence is possible:
* EOS event is pushed to queue element and still buffers are queued
* During draining remaining buffers, reconfiguration downstream
happens due to track switch.
* The queue gets a not-linked flow return from downstream
* Because the sinkpad is EOS, the queue registers an
error on the bus, causing the pipeline to fail.
Avoid the sinkpad getting marked EOS in the first place, by using a
custom event in place of EOS.
https://bugzilla.gnome.org/show_bug.cgi?id=777009
When shutting down decodebin2 and parsebin, they set their
output pads to flushing, and there is a very small window
where elements might send a sticky event such as a tag event
(which silently fails due to flushing) and then sends a buffer,
and the buffer will return GST_FLOW_ERROR because it can't
forward sticky events. The element will then send an error
message on the bus. This can also happen when elements send EOS
just as shutdown is happening. Since we're about to destroy all
the elements inside parsebin and decodebin anyway, just discard
error messages from them.
A nicer but more difficult fix for GStreamer 2.0 is to make
all event pushing / handling in core return a GstFlowReturn
like buffers do, so we can report a FLUSHING state cleanly.
Make sure ticks start with an accumulator value of 0 by incrementing it
after filling in samples instead of before and by resetting the accumulator
every time a tick begins. This prevents it from being discontinuous at the
beginning of the tick.
https://bugzilla.gnome.org/show_bug.cgi?id=774050
When plugging and then exposing a parser, don't fail
if it fails to send sticky events. The most likely
reason is that things were flushed due to the app
immediately doing a seek, but we can't detect flushing
separately to other error conditions without a
gst_pad_send_event_full() core function that returns
a GstFlowReturn.
In some case we might have EncodingProfile that will be defined
in a way that, for example if a Preset is not present, another
profile for that stream should be used.
A test is added showing the feature.
https://bugzilla.gnome.org/show_bug.cgi?id=776188
There are cases when there is no demuxer involved that could do the
buffering, e.g. HLS with raw MP3 or AAC. In this case we want to place
the buffering multiqueue after the parser.
Before this change, we've considered the first element after the
adaptive streaming demuxer as a parser. This is not always true, e.g.
id3demux. Instead we now wait until we actually have a parser (or
decoder).
Fixes playback on such HLS streams.
