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
The g-i stuff for this helper lib was never usable from bindings
anyway and there are problems with the latest gobject-introspection,
so we might just as well remove the g-i integration entirely for
this lib.
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
When dealing with streams/contents which have large duration, it is
more user-friendly to show more details in the high values (hours or days)
than in the microseconds.
This patch will use the following formatting schemes:
* Below 1hour : MM:SS.SSS
* Below 24hours : HHhMMmSSs
* Above : DDdHHhMMm
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
In GStreamer 1.12 and older, the GstBaseSrc live lock used to be held while
create() virtual function was called. As appsrc pushes serialized event in
that virtual function, we ended up with some deadlock while setting the
state to NULL. This test simulates this situation.
https://bugzilla.gnome.org/show_bug.cgi?id=783301
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
... as expected later on when end time is used to determine end running time.
Otherwise the latter is determined as NONE and the resulting text buffer is
then used indefinitely.
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.
