92a4cfe20f
When using qtdemux in a pipeline that should only work as a pure demuxer (not for actual playback), qtdemux shouldn't emit new GstSegments to correct the start time (jump to the future) to ensure that the user experiences no playback delay. By doing so, it's generating the wrong segments when an append of data from the past happens. When that happens, downstream elements such as parsers (eg: aacparse) may clip those buffers laying before the GstSegment and create problems on the GStreamer client app (eg: WebKit). Getting buffers clipped out because of the wrong GstSegments started becoming a problen when this commit was introduced: ab6e49e9cc audioparsers: add back segment clipping to parsers that have lost it This clipping makes test DASH shaka 35 from MVT tests[1] to fail in WebKitGTK/WPE (at least) and can potentially cause a number of other problems in the WebKit Media Source Extensions (MSE) code. Note that this new behaviour of not emitting new GstSegments only makes sense when qtdemux is being used as a pure demuxer and not as part of a regular pipeline. This is why the variant field has been added. When equal to VARIANT_MSE_BYTESTREAM, it will make qtdemux behave differently in push mode, taking decisions that meet the expectations for an MSE-like processing mode. This kind of tweaks have been done in the past for MSS streams, for instance. That code has been refactored to use VARIANT_MSS_FRAGMENTED now, instead of its own dedicated boolean flag. Co-authored by: Alicia Boya García <ntrrgc@gmail.com> ...who suggested to use "variant: mse-bytestream" in the caps to identify that mode, as proposed in her unmerged patch: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/467 [1] https://github.com/rdkcentral/mvt Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3867>
This the official GStreamer integration testsuites implementation repository.
Those testsuites are meant to be used with GstValidate available in the
gst-devtools[0] repository
This repository includes both media files (available in medias/)
that will be retrieved from the web using medias/get_files.py and several
testsuites (available in testsuites/) to be used with gst-validate-launcher[1].
By default, gst-validate-launcher --sync will retrieve that repository to run
the default testsuite (namely validate and ges testsuites from testsuites/validate.py
and testsuites/ges.py)
For any USAGE documentation, you should refer to gst-validate documentation and
in particular to:
$ gst-validate-launcher --help
Note that this repository uses [git-lfs] so you will need to install it before cloning.
[0] http://cgit.freedesktop.org/gstreamer/gst-devtools/
[1] https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-validate/html/gst-validate-launcher.html
[git-lfs] https://git-lfs.github.com/