mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-29 11:40:38 +00:00
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:
|
||
|---|---|---|
| .. | ||
| appsrc | ||
| aspectcropratio | ||
| baseparse | ||
| clock_sync | ||
| dash | ||
| decryptor | ||
| flow | ||
| flvdemux | ||
| h264 | ||
| h265parse | ||
| interlace | ||
| matroska | ||
| mp4 | ||
| mse | ||
| nle | ||
| playbin | ||
| playbin3 | ||
| rtp | ||
| scaletempo | ||