With playbin the last subtitle chunk would not get displayed
if the last chunk was missing a newline at the end. This is
because streamsynchronizer will hold back the EOS event until
the audio and video streams are finished too, so subparse
would never forcefully push out the last chunk until the very
end when it is too late.
We get a STREAM_GROUP_DONE event from streamsynchronizer however,
so handle that like EOS and force out any remaining text then.
https://bugzilla.gnome.org/show_bug.cgi?id=771853
(yes, this has never worked since it was introduced, don't worry)
If we want to actually detect layer/channels/samplerate changes,
it would be better to:
* not reset the various prev_* variables at every iteration.
* and actually store the values when they change
CID #206079
CID #206080
CID #206081
To passthrough crop-meta, the converter would need to allocate and
convert buffers of the size of the originating buffer. This is currently
made difficult by GstBaseTransform since we cannot alter the caps passed
though the allocation query. We would also need to wait for the first
input buffer to be received in order to make the decision around that
size.
So the short and safe solution is just to stop pretending we can
passthrought that meta.
https://bugzilla.gnome.org/show_bug.cgi?id=791412
If select-stream event was send to playbin3 as missing any GstStream of ES type
(V or A or TEX) of collection then, playbin will access to invalid address of
GstStream due to invalid index limit. This caused SIGSEGV.
https://bugzilla.gnome.org/show_bug.cgi?id=791638
The qt typefinder uses guint64 values for offset and size calculation
but the typefinder system only supports gint64 values.
Make sure we don't end up using potentially overflowing values.
The qt typefinder uses guint64 values for offset and size calculation
but the typefinder system only supports gint64 values.
Make sure we don't end up using potentially overflowing values.
n_frames could end up being quite big (potentially up to G_MAXINT64). Which
would result in overflowing 64bits when multiplying it by GST_SECOND.
Instead move GST_SECOND to the num argument
If we are shutting down, don't spawn a cleanup thread to cleanup old
groups and instead queue them to be cleaned up in the state change
thread.
This avoids (hopefully for good) having a race between the state change
thread and other threads trying to deactivate elements/pads.
Deactivating pads from two threads isn't 100% MT-safe. There is a
slim chance that the GstPadActivateFunc might be called twice with
the same values (in this case from the cleanup thread *and* from
the GstElement change_state function when going from PAUSED to READY).
In order to avoid that, call any existing cleanup function *before*
calling the parent change_state implementation on downwards state
changes.
When deactivating pads, we need to ensure that the streaming threads
going through the pads we wish to deactivate can cleanly return.
Failure to do that would result in the streaming locks of those
pads never being released. The end result would be a deadlock
when stopping decodebin2.
In order to avoid that situation, release the "dyn" lock around
the deactivation code. And refactor the code to cope with the
list of blocked pads having potentially changed when re-acquiring
the lock.
We have a dedicated one-shot thread to handle cleanup of old groups.
While this is a good idea. It's an even better idea to make sure
that thread is *completed* before the parsebin element to which
it is related isn't freed/gone.
* There can only be one cleanup thread happening at any point in time.
If there is already one, we wait for the previous one to finish.
* When shutting down (NULL=>READY) make sure the thread is finished
https://bugzilla.gnome.org/show_bug.cgi?id=790007
We have a dedicated one-shot thread to handle cleanup of old groups.
While this is a good idea. It's an even better idea to make sure
that thread is *completed* before the decodebin2 element to which
it is related isn't freed/gone.
* There can only be one cleanup thread happening at any point in time.
If there is already one, we wait for the previous one to finish.
* When shutting down (NULL=>READY) make sure the thread is finished
https://bugzilla.gnome.org/show_bug.cgi?id=790007
Instead of emitting 'drained' whenever every single chain is drained
(which would result in plenty of signal emission, and would also
occur when switching groups), only emit it when the top-level chain
is drained.
Furthermore, mark unknown (and therefore unexposed) pads as drained
since we'll never get EOS on them.
https://bugzilla.gnome.org/show_bug.cgi?id=787367
If we can expose the main chain, recheck whether we are shutting
down or not.
decodebin2 might have been set to READY/NULL during the attempt
to expose, which would cause it to fail ... but it is not a fatal
issue.
By select-streams event, current implementation of decodebin3
supports deactivate output stream (i.e., decoder element)
in reassign slot(), but cannot activate any slot without track change.
https://bugzilla.gnome.org/show_bug.cgi?id=778015
Application might choose only specific type among all available types
using select-streams event. In this case, it is desired that reconfigure
of playsink to clear unused stream path.
https://bugzilla.gnome.org/show_bug.cgi?id=778015
When an empty mix matrix is passed, audio-channel-mixer
will now generate a (potentially truncated) identity matrix,
this replicates the behaviour of audiomixmatrix in first-channels
mode.
https://bugzilla.gnome.org/show_bug.cgi?id=788833
remove_format_info was a bit confusing to read, this removes
it in favor of standard gst_caps_map_in_place calls.
This no longer simplifies the resulting caps, but I
consider this should be the job of basetransform.
https://bugzilla.gnome.org/show_bug.cgi?id=785471
Use the intended sequence for re-using elements:
* EOS
* STREAM_START if element is to be re-used
This avoids having elements (such as queue/multiqueue/queue2) not
properly resetting themselves.
When delaying EOS propagation (because we want to wait until all
streams of a group are done for example), we re-trigger them by
first sending the cached STREAM_START and then EOS (which will
cause elements to re-set themselves if needed and accept new
buffers/events).
https://bugzilla.gnome.org/show_bug.cgi?id=785951
It is forwarding messages to the playbin bus, thus forwarding messages
that contain a floating reference to the application. This generally
makes bindings unhappy, we must not leak floating references to them.
channels=1 is always mono, having it 'unpositioned' does not make
sense.
This fixes pipeline such as:
gst-validate-1.0 audiotestsrc ! audio/x-raw,channels=2,rate=44100,layout=interleaved ! audioconvert ! audioresample ! audio/x-raw, rate=44100, channels=1 ! avenc_mp2 ! fakesink
https://bugzilla.gnome.org/show_bug.cgi?id=785407
Do not remove other parsebin's input streams. It will cause unexpected
removal of any input streams in multi-parsebin use case.
Basically, the purpose of blocking buffers is similar to checking
no-more-pads of chain/group. That is, it gives hint to know the timing
to remove old (EOSed) streams of the parsebin and to add/reuse slots
for new input streams. But, that doesn't mean that we need to remove
other parsebin's EOSed stream. Each parsebin has most likely its
own streaming thread and therefore EOSed time can be much different.
(i.e., much early EOS of subtitle only parsebin)
https://bugzilla.gnome.org/show_bug.cgi?id=785120
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
