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
Signal sparse streams properly in stream-start event and force sending
of pending sticky events which have been stored on the pad already and
which otherwise would only be sent on the first buffer or serialized
event (which means very late in case of subtitle streams). Playsink in
playbin waits for stream-start or another serialized event, and if we
don't do this it will wait for the multiqueue to run full before
starting playback, which might take a couple of seconds.
https://bugzilla.gnome.org/show_bug.cgi?id=734040
All pads of a stream are now added at the beginning. In order to cope with
streams that don't get any data (forever or for a long time) we detect gaps
and push out GAP events when needed.
Cleanups and commenting by Jan Schmidt <jan@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=734040
Some VC1 decoder can have different caps according to wmv format, ie
WMV3 or WVC1.
So instead of keeping the first available caps, we interserct with
current WMV format.
https://bugzilla.gnome.org/show_bug.cgi?id=738532
When stream-format is ASF or sequence-layer-raw-frame, we basically have
a raw frame so we can parse it to extract some information such the
keyframe flag. The only requirement is to have a valid sequence-header.
This commit parse the frame header and set the DELTA_UNIT buffer flag in
case the frame is not a keyframe.
https://bugzilla.gnome.org/show_bug.cgi?id=738519
frame-layer header is represented as a sequence of 32 bit unsigned
integer serialized in little-endian byte order, so framesize is on the
first 3 bytes.
SMPTE 421M Annex L.
https://bugzilla.gnome.org/show_bug.cgi?id=738243
