Merge branch 'master' into 0.11

[platform/upstream/gstreamer.git] / gst-libs / gst / audio / gstbaseaudiosink.c

diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index bd2c601..6d09d12 100644 (file)
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -704,6 +704,7 @@ gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
   GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
   GstRingBufferSpec *spec;
   GstClockTime now;
+  GstClockTime crate_num, crate_denom;
 
   if (!sink->ringbuffer)
     return FALSE;
@@ -745,6 +746,13 @@ gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
     gst_ring_buffer_activate (sink->ringbuffer, TRUE);
   }
 
+  /* due to possible changes in the spec file we should recalibrate the clock */
+  gst_clock_get_calibration (sink->provided_clock, NULL, NULL,
+      &crate_num, &crate_denom);
+  gst_clock_set_calibration (sink->provided_clock,
+      gst_clock_get_internal_time (sink->provided_clock), now, crate_num,
+      crate_denom);
+
   /* calculate actual latency and buffer times.
    * FIXME: In 0.11, store the latency_time internally in ns */
   spec->latency_time = gst_util_uint64_scale (spec->segsize,