Performing a gst_sdp_media_get_caps_from_media() would result in
changing fields in the GstSDPMedia violating the const tag in the
function declaration.
Before there would be a line with a=rtpmap:96 VP8/90000
after, that attribute would only contain a=rtpmap:96
Fix by performing modifications on duplicated strings instead of on
the internal values.
Also add a simple test for checking that the representation doesn't
change by a gst_sdp_media_get_caps_from_media()
HLS files can have arbitrary extra tags in them, and
those can be quite long lines. We need to search
further than 256 bytes sometimes just to get past the
first few lines of the file. Make the limit 4KB,
which matches a typical input block size and should
hopefully cover every crazy input.
https://bugzilla.gnome.org/show_bug.cgi?id=780559
The path was only adding the build root. We need to also add the
prefix for the case we work with installed setup. As the search is
recursive, I had to remove any subdirectory to the already present build
root.
The term stride is confusing here, since the stride is always use
to signal the pixel row size of an image (including padding). Also
a frame may have a single stride, which adds to the confusion. This
patch uses frame-size, which simply indicate the frame size in the
case the images have some padding in between.
https://bugzilla.gnome.org/show_bug.cgi?id=780053
This allow using those property through gst-launch-1.0. This type
gained a deserilizer recently. The syntax is: <val1, val2, ...>.
Note that we also use the type int instead of uint to avoid having
to cast when specifying the values. The deserilizers assume
int by default.
https://bugzilla.gnome.org/show_bug.cgi?id=780053
When a clip has video audio and subtitle, if need send gap event
to audio and subtitle, we should make sure all has been sent, so
need every stream keep one send_gap_event.
https://bugzilla.gnome.org/show_bug.cgi?id=780429
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.
