2 seconds might be too small for some container formats, e.g.
MPEGTS with some video codec and AAC/ADTS audio with 700ms
long buffers. The video branch of multiqueue can run full while
the audio branch is completely empty, especially because there
are usually more queues downstream on the audio branch.
Usually these buffers are multiple seconds large, and having a maximum
of 5 buffers in the multiqueue there can use a lot of memory. Lower
this to 2 for adaptive streaming demuxers.
The typefinder returns LIKELY for as little as one possible
sync and no bad sync (not even taking into account how much
data was looked at for that). It's generally just not fit
for purpose, so should just not return anything like LIKELY
at all ever, even more so since it only recognises one out
of ten H263 files, and likes to mis-detect mp3s as H263.
https://bugzilla.gnome.org/show_bug.cgi?id=700770https://bugzilla.gnome.org/show_bug.cgi?id=725644
If we have the peer caps and a caps filter, return peer_caps +
intersect_first (filter, converter_caps) instead of
intersect_first (filter, peer_caps + converter_caps) and preservers
downstream caps preference order.
https://bugzilla.gnome.org/show_bug.cgi?id=724893
If we are using an adaptive stream demuxer, which outputs a non-container
stream, we are putting another multiqueue after the *parser* following
the adaptive stream demuxer. We do not want to add another instance of
the same parser right after this multiqueue.
Otherwise we will emit buffering messages not just from the last
multiqueue but also from previous multiqueues... confusing the
application with different percentages during pre-rolling.
For adaptive streaming demuxer we insert a multiqueue after
this demuxer. This multiqueue will get one fragment per buffer.
Now for the case where we have a container stream inside these
buffers, another demuxer will be plugged and after this second
demuxer there will be a second multiqueue. This second multiqueue
will get smaller buffers and will be the one emitting buffering
messages.
If we don't have a container stream inside the fragment buffers,
we'll insert a multiqueue below right after the next element after
the adaptive streaming demuxer. This is going to be a parser or
decoder, and will output smaller buffers.
Adaptive streams should download its data inside the demuxer, so
we want to use multiqueue's buffering messages to control the
pipeline flow and avoid losing sync if download rates are low;
https://bugzilla.gnome.org/show_bug.cgi?id=707636
Otherwise there's an interesting race condition when we destroy
the inputselector (actually it will be destroyed later when its state
change message gets destroyed) and afterwards release its sinkpad.
This is the code path when the last channel is removed from the
input selector.
Gave this warning sometimes, for chained oggs or whenever else
we change decode groups:
GStreamer-CRITICAL **: Padname '':sink_0 does not belong to element inputselector0 when removing
MONO and NONE position are the same, for example, but in
general there isn't much to do here for such a conversion.
Fixes problem in audioconvert, which would end up using
a mixmatrix when converting between different mono format
because it thinks MONO positioning is different from
unpositioned channels, which is not the case in this
special case. The mixmatrix would end up being 0.0 so
audioconvert would convert to silence samples.
https://bugzilla.gnome.org/show_bug.cgi?id=724509
If the text pads does not go away we just set the overlay to silent, which
allows us to immediately re-enable subs later again. However before this
change we also released the streamsynchronizer text pads, which deadlocked
because there was still dataflow going on. Just do this only if we remove
the complete chain.
https://bugzilla.gnome.org/show_bug.cgi?id=683504
Change the way autoplug-select is accumulated so that it's possible to have
multiple handlers. The handlers keep getting called as long as they keep
returning GST_AUTOPLUG_SELECT_TRY.
One practical example of when this is needed is when hooking into playbin's
uridecodebin, which is perhaps not very elegant but the only way to influence
which streams playbin autoplugs/exposes.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723096
Discussion on IRC indicated that the main reason for this list was to
prevent demuxers that can trigger a lot of seeking from using
progressive buffering using queue2 (which due to being seekable triggers
that behaviour).
However given that upstream can indicate seeks are possible but should
be avoided via a scheduling query, this extra whitelisting shouldn't be
necessary for well-behaved demuxers.
https://bugzilla.gnome.org/show_bug.cgi?id=704933
Make a little table of conversions and manually score them. Use this
info to define better weights for the scoring algorithm.
give separate scores for doing changes and the impact of the change,
This allows us to avoid conversion when we can but still allow fairly
lossless changes.
The old code did not penalize GRAY conversions, PAL conversions were
punished too low and depth conversions too high.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722656
Don't try to interpolate the chroma samples, the used algorithm only
works for horizontal cositing. Let's switch to a faster and safer
version until we handle chroma siting correctly in the fastpaths.
Rework the orc code to be around 10% faster and support arbitrary matrices.
Pass the matrix parameters to the YUV->RGB functions to make them work
for all matrices. This enables more and faster fastpath conversions.
See https://bugzilla.gnome.org/show_bug.cgi?id=721701
This fast-path was adding 128 to every component including
alpha while it should only be done for all components except
alpha. This caused wrong alpha values to be generated.
Also remove the high-quality I420 to BGRA fast-path as it needs
the same fix, which causes an additional instruction, which causes
orc to emit more than 96 variables, which then just crashes.
This can only be fixed in orc by breaking ABI and allowing more
variables.
