This commit add an helper to convert a frame to frame-layer format and
use it to implement these two stream-format conversion:
- asf --> sequence-layer-frame-layer
- asf --> frame-layer
In simple/main profile, we basically have a raw frame, so building a
frame layer isn't too complicated. But in advanced profile, the first
frame-layer should contain sequence-header, entrypoint, and frame and
each keyframe should contain entrypoint, so we have to handle these
carefully.
https://bugzilla.gnome.org/show_bug.cgi?id=738526
Add an helper to check that output stream-format is coherent with
profile and header-format. It also check if we know how to do the
conversion if the input stream-format differs from selected
output-format.
So, in case output stream-format is not allowed, it will now fail at
negotiation rather than in pre_push_frame.
https://bugzilla.gnome.org/show_bug.cgi?id=738526
This commit introduces an helper to convert an ASF frame to BDUs format with
startcodes and use this helper to implements following stream-format
conversions:
- asf --> bdu
- asf --> sequence-layer-bdu
- asf --> sequence-layer-raw-frame
https://bugzilla.gnome.org/show_bug.cgi?id=738526
It add the support of following stream-format conversion:
- bdu --> sequence-layer-bdu
- bdu-frame --> sequence-layer-bdu-frame
- frame-layer --> sequence-layer-frame-layer
For these conversion, the only requirements is to push a sequence-layer
buffer prior to data.
https://bugzilla.gnome.org/show_bug.cgi?id=738526
It prepares the template for stream-format conversion and it implements
the following conversion:
- sequence-layer-bdu --> bdu
- sequence-layer-bdu-frame --> bdu-frame
- sequence-layer-frame-layer --> frame-layer
Work is done in the pre_push_frame() method.
https://bugzilla.gnome.org/show_bug.cgi?id=738526
gstinteraudiosrc.c: In function 'gst_inter_audio_src_create':
gstinteraudiosrc.c:339:27: error: variable 'buffer_samples' set but not used [-Werror=unused-but-set-variable]
guint64 period_samples, buffer_samples;
^
The whole not_linked optimisation is really a bit dodgy here, but
let's leave it in place for now and at least start pushing data
again when a pad got linked later, in which case we should get a
RECONFIGURE event.
Current CLAMP checks both if the value is below 0 or above 255. Considering it
is an unsigned value it can never be less than zero, so that comparison is
unnecessary. Switching to using if just for the upper bound.
CID #1139796
Value from left_luminance is assigned to out_luminance here, but that stored
value is not used before it is overwritten in the next cycle of the loop.
Removing assignation.
CID #1226473
As a consequence, tsdemux won't remove its pads anymore on EOS.
Fixes the case when mpegtsbase is not able to process new packets
after EOS as the corresponding pids aren't known anymore because
the programs were removed and the pes/psi were kept, preventing the
PAT to be parsed again.
https://bugzilla.gnome.org/show_bug.cgi?id=738695
It was using a 24000/24000/48000, but I think it meant to use
24000/32000/48000. Not 100% sure...
https://en.wikipedia.org/wiki/G.722.1 has the list of supported
bitrates. It's not clear whether the "flag" code maps to this,
however.
Coverity 206072
This parses the frame_packing_arragement() payload in SEI message.
This information can be used by decoders to appropriately rearrange the
samples which belong to Stereoscopic and Multiview High profiles.
https://bugzilla.gnome.org/show_bug.cgi?id=685215
Signed-off-by: Sreerenj Balachandran <sreerenj.balachandran@intel.com>
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Assume that small backward PCR jumps are just from upstream packet
mis-ordering and don't reset timestamp tracking state - assuming that
things will be OK again shortly.
Make the threshold for detecting discont between sequential buffers
configurable and match the smoothing-latency setting on tsparse
to better cope with data bursts.
When the set-timestamps property is set, use PCRs on the provided
(or autodetected) pcr-pid to apply (or replace) timestamps on the
output buffers, using piece-wise linear interpolation.
This allows tsparse to be used to stream an arbitrary mpeg-ts file,
or to smooth jittery reception timestamps from a network stream.
The reported latency is increased to match the smoothing latency if
necessary.
Otherwise a magic capsfilter after the source is required with
exactly the same caps as the input.
This would've failed before with invalid buffer sizes:
gst-launch-1.0 videotestsrc ! intervideosink intervideosrc ! "video/x-raw,width=640,height=480" ! xvimagesink
Audiomixer blocksize, cant be 0, hence adjusting the minimum value to 1
timeout value of aggregator is defined with MAX of MAXINT64,
but it cannot cross G_MAXLONG * GST_SECOND - 1
Hence changed the max value of the same
https://bugzilla.gnome.org/show_bug.cgi?id=738845
