This provides an audio-filter and video-filter property to allow
applications to set filter elements/bins. The idea is that these will
e
applied if possible -- for non-raw sinks, the filters will be skipped.
If the application wishes to force the application of the filters, this
can be done by setting the new flag introduced on playsink -
GST_PLAY_FLAG_FORCE_FILTERS.
https://bugzilla.gnome.org/show_bug.cgi?id=679031
This provides an audio-filter and video-filter property to allow
applications to set filter elements/bins. The idea is that these will be
applied if possible -- for non-raw sinks, the filters will be skipped.
If the application wishes to force the application of the filters, this
can be done by setting the new flag introduced on playsink -
GST_PLAY_FLAG_FORCE_FILTERS.
https://bugzilla.gnome.org/show_bug.cgi?id=679031
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.
If a pipeline fails to preroll, it might happen that the sinks are
put into READY state from playbin's sink activation, but they are never
set to playsink, so they aren't being managed by a GstBin and will keep
their READY state until they are unreffed, leading to a warning.
Prevent this by always forcing them to NULL when deactivating a group
https://bugzilla.gnome.org/show_bug.cgi?id=708789
Fix component ordering, it's wrong in both the scanline and merge
function so it cancels eachother out and isn't really a except for
loss of precision of the green component.
Fix calculation of the filter weight
Some of the fastpath function can only work with aligned widht/height
so make sure we check this as well when choosing a fastpath.
Add fastpath for I420/YV12 -> BGRx
This commit adds detection of the "dash" and "avc3" compatible brands
in qt_type_find.
Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new
structure for fragmented MP4 called "avc3". The principal difference
between AVC1 and AVC3 is the location of the codec initialisation
data (e.g. SPS, PPS). In AVC1 this data is placed in the initial MOOV
box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data goes in
the first sample of every fragment (i.e. the first sample in each mdat
box). The principal reason for avc3 is to make it easier for client
implementations, because it removes the requirement to insert the
SPS+PPS in to the decoder pipeline every time there is a representation
change.
https://bugzilla.gnome.org/show_bug.cgi?id=702004
Increase the number of temporary lines that we need, it is possible that the
up and downsampling offsets are out of phase and that we need to keep some
extra lines around. Also copy the unhandled output lines for the next round
instead of overwriting them.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706823
When playing mp3 files from a smb server, we get 64k read requests
that mostly overlap. Without using the cache to partially satisfy
these, we send these requests straight to the server, resulting in
a lot more network traffic than necessary.
https://bugzilla.gnome.org/show_bug.cgi?id=705415
Each write will update the last_activity_time and otherwise we would
compare against a too old current time and immediately timeout because
current time is smaller than last activity time (overflow).
Each write will update the last_activity_time and otherwise we would
compare against a too old current time and immediately timeout because
current time is smaller than last activity time (overflow).
Remove dodgy code that detects mp3 with as little as
a valid frame sync at the beginning. This was only used
in some unit tests in -good where there were only a few
bytes after the id3 tag. We now require at least two
frame headers.
Fixes mis-dection of text files with UTF-16 LE BOM as mp3.
https://bugzilla.gnome.org/show_bug.cgi?id=681368
We have to hold the streams-lock when iterating over all pads,
also the stream-lock of the pad is already locked when we receive
EOS.
Call gst_pad_event_default() for the correct default handling of
events.
