gstreamer/gst-libs
Wim Taymans 95d162fb71 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Change the way in which the ringbuffer is started when dealing with a
slaved clock and latency. We now sync to the clock until we reach
upstream latency before starting the ringbuffer. This has the effect
that we can accurately align the master and slave clocks and let the
rate correction code take care of the initial drift or rounding errors
instead of leaving them uncorrected with the old approach.
2008-05-20 11:09:06 +00:00
..
ext move ffmpeg stuff to gst-ffmpeg module 2004-02-13 15:11:50 +00:00
gst gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c... 2008-05-20 11:09:06 +00:00
Makefile.am configure.ac: added GST_LIB_LDFLAGS and GST_ALL_LDFLAGS 2005-11-27 16:18:50 +00:00