There are broken(?) mjpeg videos that are incorrectly detected as
interlaced. This happens because 'info.height > height' (e.g. 1088 > 1080).
In the interlaced case info.height is approximately 'height * 2' but not
exactly because height is a multiple of DCTSIZE. Make the check more
restrictive but take the rounding effect into account.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4696>
For interlaced jpeg, gst_jpeg_dec_decode_direct() is called twice, once for each
field. In this case, stride[n] is plane_stride[n] * 2 to ensure that only every
other line is written. So the loop must stop at height / num_fields.
If the frame is really interlaced then continuing beyound this, is not harmful,
because jpeg_read_raw_data() will do nothing and return 0, so am info message is
printed.
However, if the frame is not actually interlaced, just misdetected as interlaced
then there is still data available from the second half of the frame. Now
line[0][j] is set to the scratch buffer. If the scratch buffer is not allocated
(because the height is a multiple of v_samp[0] * DCTSIZE) then the result is a
segfault due to a null-pointer dereference.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4696>
Assigning TRUE (1) to a signed 1 bit integer will cause truncation
from 1 to -1 because the only non-zero value that can be stored is -1
due to how two's-complement works.
As this is a proper GObject let's not bother with all this and simply
use a normal gboolean instead.
../subprojects/gst-plugins-good/ext/pulse/pulsesink.c:1490:19: warning: implicit truncation from 'int' to a one-bit
wide bit-field changes value from 1 to -1 [-Wsingle-bit-bitfield-constant-conversion]
pbuf->in_commit = TRUE;
^ ~~~~
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4617>
The qt5 and qt6 plugins will now correctly error out if you enable the
option, and you can also now explicitly ensure that wayland, x11,
eglfs support is actually functional by enabling the options. It was
too easy to build non-functional support for these.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4537>
jackaudiosink and jackaudiosrc have a rank and might be plugged
as part of auto-plugging inside playbin and playsink or the
autoaudiosink/autoaudiosrc elements, so we don't really want to
spew ERROR log messages in that case, which is consistent with
what alsasink and pulseaudiosink do.
This is less noticable on Linux because pulseaudiosink has a
higher and alsasink which has the same rank comes before jack
in the alphabet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4545>
In order to provide build and provide the jack plugin with the prebuilt
binaries of gstreamer we distribute with releases, we can not depend
on an external dependency nor can we ship plugins linking to libraries
we don't provide.
We can also not provide jack ourselves, as it would likely cause a
mismatch with the jack daemon on the host.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4350>
Short-circuit parsing and recreating the playlist URI if
no HLS directives are going to be applied to it.
Fixes problems playing some streams (YouTube) that have
unneeded escaped characters in the URI and then complain
when GStreamer removes the escaping
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4335>
There's no guarantee it will *actually* be the URI which refered to what we are
downloading. It could be a stream URI or anything else.
Instead of putting something wrong, put no (specific) referer as a better choice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3972>
Makes "start-bitrate" work without setting "connection-speed" property. Having
another property set as a requirement for this one to work is unexpected.
This commit allows to request some initial bitrate for first segment, then
go into adaptive streaming for the rest of media playback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3895>
The live playlists should be updated at a defined interval. The problem is that
this interval was used *after* the playlist was finally received and processed,
which resulted in a gradual shift happening in playlist updates.
Instead store and use the time at which playlists were requested to determine
when the next one should be downloaded.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
The scanning is done in a reverse order, the proper full checks to do are
therefore:
* If the position is beyond half a "segment duration", it's in the following
segment
* If the position is within the first half of a segment, it's in that one
* If the segment is the first one and the position is within half a duration
backwards, we consider the position as being within that first segment
Also handle the case where a "partial only" segment doesn't have a reliable
duration, and therefore use the playlist target duration instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
The implementation wouldn't work with regular HLS streams (i.e. the final
fallback).
Now that the implementation uses time to search for the starting
segment (instead of just the n-th from the end), we can specify the correct
hold_back fallback value from the RFC
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Avoid a deadlock if a downstream seeking query happens while the scheduler
thread is holding the manifest lock (for example during a seek back to live).
Instead, do a more elaborate fix where the external calls that need access to a
'manifest' access a copy that's updated during a manually triggered manifest
update callback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
Rename track_dequeue_data_locked() to
gst_adaptive_demux_track_dequeue_data_locked(), since it's non-static.
Make find_stream_for_track_locked() static since it's only used in the main
gstadaptivedemux.c file.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
gst_adaptive_demux2_stream_finish_download() will already schedule another
fragment download if it can so don't fall through to the retry code that will
also try and schedule a download (triggering an assert).
Fix the logic in general to retry advancing into the live seek range once.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
When calculating the seek range for a live stream, use the same hold-back logic
as when choosing a starting segment, including low-latency segments if
enabled. Permits seeking closer to the live edge when re-synching or catching
up.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
