[0.11] gst-plugins-base: audiosink: improve comment

[Wim Taymans]

Module: gst-plugins-base
Branch: 0.11
Commit: d10602fbdece5bf83e74362639c3bd061a63f305
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-base/commit/?id=d10602fbdece5bf83e74362639c3bd061a63f305

Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date:   Mon Mar 28 10:25:38 2011 +0200

audiosink: improve comment

---

 gst-libs/gst/audio/gstbaseaudiosink.c |    5 ++++-
 1 files changed, 4 insertions(+), 1 deletions(-)

diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index 30d9b73..3508382 100644
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -739,7 +739,10 @@ gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
 
   GST_DEBUG_OBJECT (sink, "release old ringbuffer");
 
-  /* get current time, updates the last_time */
+  /* get current time, updates the last_time. When the subclass has a clock that
+   * restarts from 0 when a new format is negotiated, it will call
+   * gst_audio_clock_reset() which will use this last_time to create an offset
+   * so that time from the clock keeps on increasing monotonically. */
   now = gst_clock_get_time (sink->provided_clock);
 
   GST_DEBUG_OBJECT (sink, "time was %" GST_TIME_FORMAT, GST_TIME_ARGS (now));