gst-plugins-base: baseaudiosink: commit correct number of samples when not syncing

Mark Nauwelaerts mnauw at kemper.freedesktop.org
Tue Jan 17 12:49:36 PST 2012


Module: gst-plugins-base
Branch: master
Commit: 3e312e6e162638d8e07f0edb3859980dabb089da
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-base/commit/?id=3e312e6e162638d8e07f0edb3859980dabb089da

Author: Mark Nauwelaerts <mark.nauwelaerts at collabora.co.uk>
Date:   Tue Jan 17 21:46:58 2012 +0100

baseaudiosink: commit correct number of samples when not syncing

---

 gst-libs/gst/audio/gstbaseaudiosink.c |    5 ++---
 1 files changed, 2 insertions(+), 3 deletions(-)

diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index e7ff30d..03e332d 100644
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -1654,7 +1654,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
         GST_BUFFER_SIZE (buf), render_start);
     /* we don't have a start so we don't know stop either */
     stop = -1;
-    goto no_sync;
+    goto no_align;
   }
 
   /* let's calc stop based on the number of samples in the buffer instead
@@ -1730,7 +1730,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
     render_stop = render_start + samples;
     GST_DEBUG_OBJECT (sink,
         "no sync needed. Using render_start=%" G_GUINT64_FORMAT, render_start);
-    goto no_sync;
+    goto no_align;
   }
 
   /* bring buffer start and stop times to running time */
@@ -1856,7 +1856,6 @@ no_align:
   /* number of target samples is difference between start and stop */
   out_samples = render_stop - render_start;
 
-no_sync:
   /* we render the first or last sample first, depending on the rate */
   if (bsink->segment.rate >= 0.0)
     sample_offset = render_start;



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