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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:
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.. | ||
ges | ||
medias@2ec828b7e7 | ||
testsuites | ||
.gitignore | ||
HACKING | ||
integration-testsuites.config | ||
meson.build | ||
README |
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/