Jan Schmidt 45e06fe704 gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an... ... Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps), (gst_ring_buffer_samples_done): * gst-libs/gst/audio/gstringbuffer.h: Document better the fact that latency_time and buffer_time are values stored in microseconds, and not the usual GStreamer nanoseconds. Change the variables (compatibly) that store them from GstClockTime to guint64 to make it more clear that they're not storing clock times. Also, remove the bogus property description that says the user can specify -1 to get the default value, since that's never been the case. When computing the default segment size for the ring buffer, make it an integer number of samples. When the sub-class indicates a delay greater than the number of samples we've written return 0 from the audio sink get_time method.		2006-06-03 21:06:49 +00:00
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ext	move ffmpeg stuff to gst-ffmpeg module	2004-02-13 15:11:50 +00:00
gst	gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...	2006-06-03 21:06:49 +00:00
Makefile.am	configure.ac: added GST_LIB_LDFLAGS and GST_ALL_LDFLAGS	2005-11-27 16:18:50 +00:00