mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-19 08:11:16 +00:00
317bb22aca
Original commit message from CVS: * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain): Don't rely on incoming buffers offset anymore, since it is completely broken when using multiple segments. Instead convert the incoming buffers timestamp to running time, and then convert that value to the offsets. Also inform GstSegment of the last outputted stop position, which is needed if we received several segments with an unknown stop value. |
||
---|---|---|
.. | ||
adder | ||
audioconvert | ||
audiorate | ||
audioresample | ||
audiotestsrc | ||
ffmpegcolorspace | ||
gdp | ||
playback | ||
subparse | ||
tcp | ||
typefind | ||
videorate | ||
videoscale | ||
videotestsrc | ||
volume | ||
Makefile.am |