? gstbaseaudiosrc.c-bak
? gstbaseaudiosrc.h-bak
? patch
Index: gstbaseaudiosrc.c
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-base/gst-libs/gst/audio/gstbaseaudiosrc.c,v
retrieving revision 1.13
diff -u -3 -p -r1.13 gstbaseaudiosrc.c
--- gstbaseaudiosrc.c	11 Oct 2005 18:32:01 -0000	1.13
+++ gstbaseaudiosrc.c	18 Oct 2005 17:33:58 -0000
@@ -294,6 +294,7 @@ gst_base_audio_src_event (GstBaseSrc * b
     case GST_EVENT_FLUSH_STOP:
       /* always resync on sample after a flush */
       src->next_sample = -1;
+      src->next_timestamp = 0;
       gst_ring_buffer_clear_all (src->ringbuffer);
       break;
     default:
@@ -334,6 +335,10 @@ gst_base_audio_src_create (GstPushSrc * 
 
   src->next_sample = sample + samples;
 
+  GST_BUFFER_DURATION(buf) = src->ringbuffer->samples_per_seg * GST_SECOND / src->ringbuffer->spec.rate;
+  GST_BUFFER_TIMESTAMP(buf) = src->next_timestamp;
+  src->next_timestamp += GST_BUFFER_DURATION(buf);
+
   gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (psrc)));
 
   *outbuf = buf;
@@ -394,6 +399,7 @@ gst_base_audio_src_change_state (GstElem
       if (!gst_ring_buffer_open_device (src->ringbuffer))
         return GST_STATE_CHANGE_FAILURE;
       src->next_sample = 0;
+      src->next_timestamp = 0;
       break;
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       break;
@@ -412,6 +418,7 @@ gst_base_audio_src_change_state (GstElem
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       gst_ring_buffer_release (src->ringbuffer);
       src->next_sample = 0;
+      src->next_timestamp = 0;
       break;
     case GST_STATE_CHANGE_READY_TO_NULL:
       gst_ring_buffer_close_device (src->ringbuffer);
Index: gstbaseaudiosrc.h
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-base/gst-libs/gst/audio/gstbaseaudiosrc.h,v
retrieving revision 1.4
diff -u -3 -p -r1.4 gstbaseaudiosrc.h
--- gstbaseaudiosrc.h	6 Oct 2005 15:15:04 -0000	1.4
+++ gstbaseaudiosrc.h	18 Oct 2005 17:33:58 -0000
@@ -60,6 +60,9 @@ struct _GstBaseAudioSrc {
   /* the next sample to write */
   guint64	 next_sample;
 
+  /* timestamp of next sample */
+  guint64	 next_timestamp;
+
   /* clock */
   GstClock	*clock;