The GSource for dealing with timeouts in
gst_video_convert_sample_async() might be attached to a non-default
context, so we should not be using g_source_remove() on the returned ID.
The correct thing to do is to keep a reference to the actual GSource and
then call g_source_destroy() on it.
https://bugzilla.gnome.org/show_bug.cgi?id=780297
When posting 100% buffering due to removing the last
buffering element, we still need to hold the posting
lock as well, to avoid any race with other elements
that might post a buffering message at that exact
moment
Add locking, and handle EOS properly now that urisourcebin
uses custom events in place of real EOS events, so we
need to manually remove buffering messages and potentially
post 100% in that situation
Track how long it takes to generate the first buffer after a flush
as a simple measure of how efficient the decoder is at skipping /
rushing to get to the first decode.
When initializing a timecode from a GDateTime, and the remaining time
until the new second is less than half a frame (according to the given
frame rate), it would lead to the creation of an invalid timecode, e.g.
00:00:00:25 (at 25 fps) instead of 00:00:01:00. Fixed.
https://bugzilla.gnome.org/show_bug.cgi?id=779866
Use G_GUINT64_FORMAT for guint64 values.
Introduced by fcb63e77a9
Found by Alexander Larsson
gstvideodecoder.c: In function 'gst_video_decoder_have_frame':
gstvideodecoder.c:3312:51: error: format '%u' expects argument of type 'unsigned int', but argument 8 has type 'guint64 {aka long long unsigned int}' [-Werror=format=]
The expanded 4 second buffering was making radio streams that are
being delivered at real-time speeds too slow. We might need
a better plan for matching the queue2 size to incoming bitrate
in the absence of tag information or timestamping.
In uridecodebin, it used tags on the output of decodebin to
adjust the queue2 buffering, but urisourcebin doesn't have that
view - decodebin is downstream from us.
Use new message notify API and print caps and taglists in a nicer
to read way, just like gst-launch-1.0 does nowadays, without
escaping everything three times.
Fix various issues with reverse playback by clearing tracking
vars when working in reverse, and where possible using the
timestamp interpolation code to generate timestamps for
outgoing buffers. Make sure to mark things as discontinuous
only when looping backward to a new position and fix seeking
to the next page when starting.
In gst_ogg_demux_do_seek() when calculating the
keyframe time, account for a non-zero start-time
Handle a discontinuous first packet in
gst_ogg_demux_setup_first_granule() because that's pretty
normal after a seek. Also differentiate between a genuinely
truncated first packet and just bailing out early, by not using
granule = -1 as an error code.
Make the debug output logs clearer about which timestamps
are stream times (PTS) and which are ogg timestamps.
Don't guess a timestamp of the start of the segment when running
in reverse mode, as more likely it means we're discontinuous somewhere
in the middle of the segment, and we'll fix up timestamps once
the frames are decoded and reversed.
When a PTS is not set, we still want to store the rest of the
buffer information, or else we lose important things like the
duration or buffer flags when parsing.
This is a followup commit to b95725c37e
* Resetting the decoder should only happen when we get a new initialization
header (0x01) and not on the other headers
* The initialized variable only gets set to TRUE once all headers have
been parsed. Also check if the vorbis_info struct has been properly resetted
also. Failure to do that would cause vorbisdec to error if it got
two initialization header in a row (the first would configure the underlying
library and the second one would error out because it's already initialized)
https://bugzilla.gnome.org/show_bug.cgi?id=779515
