This reverts commit adc9694ed7.
No need to restrict the conversion, we can handle interlace correctly. We
basically unpack each field, then convert each field to the target colorspace
and pack and interleave each field to the target format. We also disable any
fast path that can't deal with interlaced formats.
Do not use the buffer start offset when it is invalid, otherwise a
discontinuity is detected on the next buffer, and the subtitle parser
reset and some subtitle lines are not shown.
Also remove unused next_offset field.
https://bugzilla.gnome.org/show_bug.cgi?id=693981
subtitleoverlay handles any caps, not just the ones
for which a subtitle parser/renderer exist. It will
just ignore any unsupported streams instead of causing
an error.
https://bugzilla.gnome.org/show_bug.cgi?id=688476
Add all the caps that we can convert to to the filter caps,
otherwise downstream might just return EMPTY caps because
it doesn't handle the filter caps but we could still convert
to these caps, causing us to return EMPTY caps although
conversion would be possible.
https://bugzilla.gnome.org/show_bug.cgi?id=688803
changed: gst_video_scale_set_info in gst/videoscale/gstvideoscale.c
DAR on sink side now calculated with PAR on sink side
ratio of output width/height now calculated with inverse PAR
additional condition that borders are 0:0 for passthrough mode
https://bugzilla.gnome.org/show_bug.cgi?id=696019
Ensure the detection of svc and mvc as a part of h264 stream.
Once the typefinder detect a subset_sequence_parameter_set(ssps),
then each nal unit with type 14 or 20 should be detected as a
part of h264 stream thereafter.
https://bugzilla.gnome.org/show_bug.cgi?id=694346
Previously adder was only sending the flush-stop, when it saw the flushing seek.
If one sends a flushing see direcly to an element upstream of adder, it would
fail to unflush the downstream pads.
This ensures the ghost pad will not stay in flushing mode
when it receives a flush stop event, and generally behave
badly.
This fixes at least one case of a dynamic decodebin2 + encodebin
pipeline finding a source that has not prerolled when it should
have been (due to the ghostpad staying in flushing mode).
We were setting the query-func on the sink-pad, which got overwritten when
adding the new pad to collect pads. Instead register our query-func with the
collect pads object. This fixes filter caps. Add a test for it.
A return value of FALSE here indicates that we don't have control-values. In
0.10 we were returning the default value of the property. Now we don't fill an
array with defaults in the ControlBinding, but leave it up to the element to
handle this case.
The codec data blob we get from matroskademux with the SSA/ASS
init section is supposed to be valid UTF-8. If it's not, just
continue with the bits that are valid UTF-8 instead of erroring
out. We don't actually parse the init section yet anyway..
https://bugzilla.gnome.org/show_bug.cgi?id=607630
The behaviour is sensibly changed here. Instead of purely falling when a
preset is set on the #GstEncodingProfile, we now make sure that the
element that is plugged corresponds to the one specified as preset. Then,
if we have a preset_name, we use it, if it fails, we fail (we might rather
just keep working even without setting the element properties?)
+ Add tests that it behave correctly
When the input buffers for a stream don't have a duration set,
timestamp_end might still be GST_CLOCK_TIME_NONE. When advancing
EOSed streams via GAP events (with other streams not yet EOS), we
would then use the invalid timestamp_end to calculate the duration
of the gap. This in turn would make baseaudiosink abort, because it
would try to allocate memory for a trizillion samples.
So if buffers don't have a duration set, assume a duration of
one second for stream catch-up purposes, just so we can still
continue to catch up in those cases. And make sure that
timestamp_end is valid before doing calculations with it.
http://bugzilla.gnome.org/show_bug.cgi?id=678530
Make AAC LOAS typefinding a bit more reliable; don't report
a LIKELY probability already after just two sync points, but
scan for a few more consecutive frames and determine probability
based on how many we found. Fixes mis-detection of wavpack file.
https://bugzilla.gnome.org/show_bug.cgi?id=687674
Check for second block sync and return different
probabilities depending on what we found (trumping
the AAC loas typefinder's LIKELY probability after
finding a second frame sync in this particular case).
https://bugzilla.gnome.org/show_bug.cgi?id=687674
Previously we could've chosen another format with the same
depth even if the input format was possible.
Also make sure to chose according to the order in the
caps.
Enhance current code to prefer an exact match on sample depth if
possible. Also ignore GST_AUDIO_FORMAT_FLAG_UNPACK when checking
equality on the flags.
This is an adaptation of patch #3 from Jyri Sarha
( http://lists.xiph.org/pipermail/speex-dev/2011-September/008240.html ),
but without the NEON optimizations (these come in a separate commit).
The idea is to replace SATURATE32(PSHR32(x, shift), a) operations with a
combined SATURATE32PSHR(x, shift, a) macro that can be optimized for
specific platforms (and also avoids rare rounding errors).
Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
There were two issues with the previous decodebin2 group switching algorithm:
Issue 1: It operated with no memory of what has been drained or not, leading to
multiple checks for chains/groups that were already drained.
Issue 2: When receiving an EOS, it only detected that a higher-level chain
was drained if it contained the pad receiving the EOS.
The following modifications have been applied:
- a new drained property has been added to GstDecodeChain
- both drained properties of chain/group are set as soon as they are detected
- the algorithm now tests agains these values
See https://bugzilla.gnome.org/show_bug.cgi?id=685938
Should fix "cannot register existing type `GstPlaybinSelectorPad'" warnings
and subsequent errors when creating multiple players at the same time.
Conflicts:
gst/playback/gststreamselector.c
GstId3Mux sink pad is an always (static) pad. Thus releasing it
as if a request pad triggers:
(sound-juicer:11826): GStreamer-CRITICAL **:
gst_element_release_request_pad: assertion `GST_PAD_PAD_TEMPLATE (pad)
== NULL || GST_PAD_TEMPLATE_PRESENCE (GST_PAD_PAD_TEMPLATE (pad)) ==
GST_PAD_REQUEST' failed
https://bugzilla.gnome.org/show_bug.cgi?id=685110
Need to store the old running time and frame numbers when renegotiating and
start from 0 again when a new caps is set, preventing that framerate changes
cause timestamping issues.
For example, if a stream pushed 10 buffers on framerate=2/1, its
running time will be 5s. If a new framerate of 1/1 is set, it would
make the running time go to 10s as it would count those 10 buffers
as being sent on this new framerate.
Fixes camerbin unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=682973
../../../gst-plugins-base/gst/audioresample/gstaudioresample.c: In function 'gst_audio_resample_dump_drain':
../../../gst-plugins-base/gst/audioresample/gstaudioresample.c:729:9: warning: variable 'in_len' set but not used [-Wunused-but-set-variable]
